[asterisk-users] transmit_silence_during_record
I have a server that is receiving a disconnect during recording of long incoming messages. The connection is via a SIP gateway and when the gateway sees no RTP for 5 mins, it hangs up the call. I enabled transmit_silence_during_record but I see no RTP being sent from Asterisk to the gateway during the record. Is there something I need to enable besides setting transmit_silence_during_record=yes to enable some RTP traffic outbound during the record? Jonathan -- _ -- 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-users] SendFAX/T.38 question
I have some questions about the T.38 faxing capability. I have been able to successfully setup the inbound receive fax. However, I am having problems tracking down the format of the outbound extensions.conf SendFAX command. I have looked at the code and it looks like it only takes a single parameter, a file name. But the attempts I have tried seem unsucessful. I have tried dialing out and then calling SendFAX and calling SendFAX before the dial. No success. Can someone please provide me with an extensions.conf example of how to use SendFAX? Thank you. Jonathan Augenstine ___ -- 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] SendFAX/T.38 question
I have some questions about the T.38 faxing capability. I have been able to successfully setup the inbound receive fax. However, I am having problems tracking down the format of the outbound extensions.conf SendFAX command. I have looked at the code and it looks like it only takes a single parameter, a file name. But the attempts I have tried seem unsucessful. I have tried dialing out and then calling SendFAX and calling SendFAX before the dial. No success. Can someone please provide me with an extensions.conf example of how to use SendFAX? Thank you. Jonathan Augenstine ___ -- 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-users] DTMF pass-through question
I am trying to resolve an issue and I believe it is my configuration. The scenario is that I have a SIP detected on the server. The dial plan then makes a local connection to another part of the dial plan. The new dial plan extension then places another SIP call out to a SIP phone. When the call is accepted there is streamed from the calling SIP phone. When the audio is complete a DTMF is transmitted to Asterisk. The DTMF is detected by Asterisk but it does not get passed through to the other SIP phone. I would like the DTMF to pass-through to the other SIP phone. Is this a configuration issue? Or do I need to handle this on the dial plan level? Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-users] DTMF pass-through question
Matt, Asterisk version == 1.4.22 dtmfmode == info calls are bridged through Asterisk (canreinvite=no) Jonathan On Sun, Dec 28, 2008 at 3:23 PM, Matt Florell astma...@gmail.com wrote: On 12/28/08, jonathan augenstine jaugenst...@gmail.com wrote: I am trying to resolve an issue and I believe it is my configuration. The scenario is that I have a SIP detected on the server. The dial plan then makes a local connection to another part of the dial plan. The new dial plan extension then places another SIP call out to a SIP phone. When the call is accepted there is streamed from the calling SIP phone. When the audio is complete a DTMF is transmitted to Asterisk. The DTMF is detected by Asterisk but it does not get passed through to the other SIP phone. I would like the DTMF to pass-through to the other SIP phone. Is this a configuration issue? Or do I need to handle this on the dial plan level? Jonathan Asterisk version? What are both dtmfmodes set to for each SIP endpoint? Are the calls natively bridged or bridged through Asterisk? MATT--- ___ -- 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] SER, OpenSER, Kamailio, OpenSIPS -- what are you using?
Have you checked out OpenSBC (www.voip-info.org/wiki/view/*OpenSBC)?* On Fri, Dec 12, 2008 at 6:19 PM, Steve Edwards asterisk@sedwards.comwrote: One of the above is frequently used to front-end Asterisk. I used OpenSER to front-end a farm of Asterisk servers and was very happy with it. The ability to take a box out of service or to route a specific DNIS to a box for testing rocks. Since OpenSER has died (I don't care about the politics/personalities/trademarks), Kamailio and OpenSIPS have risen from the ashes. What are you using? (I'm still using OpenSER 1.3.1-notls.) Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_confcall on Asterisk 1.6 update
FYI I was informed by A. Minnesale that app_confcall was originally developed for Asterisk 1.2. He stated that there would probably be a significant amount of work to update it to Asterisk 1.6. Jonathan ___ -- 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] app_confcall build issues
I am trying to build app_confcall and it is failing. Are there known build issues with this module. I am running Asterisk 1.6.0-beta9. Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail.conf
Is it possible to create extensions in the voicemail.conf remotely by using the manager interface. I cannot seem to find any documents or examples describing that capability. Jonathan ___ -- 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] WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing
Have you verified that ztdummy is loaded? On Sun, 2006-03-26 at 01:06 -0500, Erick Perez wrote: Hi, using asterisk 1.2.5 with mysql in a centos 4.2 (2.6 kernel) no hardware interfaces installed gives me this error. Im a bit new to this so any help will be appreciated. == Parsing '/etc/asterisk/musiconhold.conf': Found Mar 26 00:58:49 WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. [chan_oss.so] = (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found == Registered channel type 'Console' (OSS Console Channel Driver) musiconhold.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 thanks, -- --- Erick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I'm FED UP with BroadVoice
I have had very reliable inbound/outbound service from Junction Networks (www.junctionnetworks.com). The one time I did have an issue, it was resolved quickly. During my testing I concluded that BroadVoice (my partner refers to them as NoVoice) was unreliable (approximately 40% of all of our test calls went uncompleted) and not ready for prime time (almost impossible to get in contact with technical support). For our inbound only service we have been using Voxbone (www.voxbone.com). Very reliable and very responsive technical support. We had one problem plaguing us that turned out to be a configuration issue that was my fault. However, they were very responsive in testing their end to help troubleshoot the problem. On Thu, 2006-03-23 at 12:05 -0700, Ronald Lewis wrote: After months of BroadVoice ignoring my trouble tickets for dropped calls, delayed termination, etc., I'm throwing in the towel. While they have credited $19.95 to my account, they refuse to credit anything more, despite ALL of the problems I've had. I feel the least they could do is credit the remaining $8.61 to my account, yet they won't. I haven't really been following up on porting between VoIP providers, but is there a remote chance I can save my phone number? I'd sure hate to change numbers again -- this has been a NIGHTMARE. Everyday, calls are dropping, and I'm calling people back 2 to 3 times to establish a decent connection. And their response (paraphrasing): We've made the best effort to ensure your service is functional ... but there are some things beyond our control with VoIP. Not good enough! I had great service with Vonage, and the times I use VoipJet, it works perfectly! Thanks in advance for any pointers. Ronald Lewis Denver, Colorado http://www.ronaldlewis.com/interviews ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about meetme app
A locked conference means that a pin number is required to join the conference. On Fri, 2006-03-17 at 16:20 -0500, Michael Gaudette wrote: I have a quick question about the MeetMe app. A locked conference means what exactly? A) That people can't join anymore B) That everyone is muted except the admin Follow-up question If the answer above is A, how do you accomplish B? Mick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Question about meetme app
My mistake. Locking a conference from the CLI does prevent any additional callers from connecting. But AFAIK locking the conference does not prevent you from muting a participant. What I was thinking in my original response was limiting a conference, not locking it, by adding a pin number. On Fri, 2006-03-17 at 17:45 -0500, Michael Gaudette wrote: As in press 2 to lock or unlock this conference in the conf admin menu? Then, how do you mute participants? I can't imagine MeetMe not having this functionality. Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Augenstine Sent: March 17, 2006 5:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Question about meetme app A locked conference means that a pin number is required to join the conference. On Fri, 2006-03-17 at 16:20 -0500, Michael Gaudette wrote: I have a quick question about the MeetMe app. A locked conference means what exactly? A) That people can't join anymore B) That everyone is muted except the admin Follow-up question If the answer above is A, how do you accomplish B? Mick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Streaming Music On Hold
Try this: musiconhold.conf: [stream2] mode=mp3 directory=http://pubint.ic.llnwd.net/stream/pubint_wnpr extensions.conf: exten = 1234,1,Answer exten = 1234,2,MusicOnHold(stream2) exten = 1234,3,Hangup On Wed, 2006-02-22 at 09:28 -0700, Douglas Garstang wrote: Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After several hours jerking around with icecast and muse, I tried to point my asterisk system directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty application=http://pubint.ic.llnwd.net/stream/pubint_wnpr and this is how I am testing it: exten = 1234,1,Answer exten = 1234,2,SetMusiconHold(stream2) exten = 1234,3,WaitmusiconHold(60) exten = 1234,4,Hangup and this is the console output I get when I dial 1234: Asterisk Ready. *CLI -- Executing Answer(SIP/3250072-ed28, ) in new stack -- Executing SetMusicOnHold(SIP/3250072-ed28, stream2) in new stack -- Executing WaitMusicOnHold(SIP/3250072-ed28, 60) in new stack -- Started music on hold, class 'stream2', on channel 'SIP/3250072-ed28' -- Stopped music on hold on SIP/3250072-ed28 If I replace SetMusiconHold(stream2) with SetMusiconHold(default), I get the default music on hold. Running ngrep on port 80 shows me that the Asterisk system is not sending or receiving ANY data on port 80. What am I doing wrong? Yes, it has network and DNS connectivity. Can't believe it's this hard! Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOH from RCA jack?
Barix Instreamer takes RCA in and MP3 or ulaw stream out. Asterisk can use either for MOH. On Fri, 2006-02-17 at 07:42 -0600, Rich Adamson wrote: Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio equipment that interfaces to their key system via a standard audio RCA jack. (We're reseaching other alternative deliver mechanisms such as mp3's, etc, from the supplier, but have to assume for now that we need to inject MOH audio into asterisk via this RCA jack.) Does anyone have a relatively high audio quanlity method of interfacing such an external audio device into asterisk in a reliable way via an RCA jack? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK Provider
You can try Voxboned(www.voxbone.com) if you need inbound only. On Tue, 2006-01-24 at 09:13 +, scott wrote: Hi Does anyone know a UK Voip Proivder that will give me more than 1 telephone number and point it to my sip account. www.SipGate.co.uk are great but they only allow 1 telephone number per user, you can register another telephone number by registering as another user but Asterisk doesn't allow multiple registrations. Many Thanks Scott ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Configuration
Here is where you will find the answer to all of your questions: http://www.asterisk.org/ http://www.voip-info.org/wiki-Asterisk Jonathan On Sat, 2005-12-24 at 01:34 +0500, Faheem Ahmed wrote: I have installed Redhat Linux 9 and Asterisk 1.2.1 on new computer. I need to know initial configuration of Asterisk i.e How to register a sip user?. What files do I have to edit? I am new about the Asterisk please help me Faheem Ahmed ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: PROGRESS with cause code 31 received
Cause No. 31 - Network disconnect (Normal, unspecified)/Special intercept announcement: Call blocked because of group restricitons It looks like a telco configuration issue. Your provider probably has toll free block on your trunk(s). You should be able to call them and ask to have it enabled. On Thu, 2005-12-15 at 21:43 -0500, Dana Olson wrote: I have upgraded to HEAD and still, this problem persists... I can dial any other numbers through my provider with no issues, but any toll-frees do NOT work still. I have searched on Google as well, and I can't find any help.. Can anyone assist? Dana On 11/28/05, Dana Olson [EMAIL PROTECTED] wrote: I have been trying to work this problem out with my IAX provider. I dial a toll-free number, ex: 1-888-876-6262, and I get a due to technical difficulties message. I set my debug level to 9, and all I see when I dial out is this: -- Executing Dial(SIP/27-51de, IAX2/voctel/1766262||T) in new stack -- Called voctel/1766262 -- Call accepted by 204.14.18.189 (format ulaw) -- Format for call is ulaw -- IAX2/voctel-3 is proceeding passing it to SIP/27-51de -- IAX2/voctel-3 is making progress passing it to SIP/27-51de -- Hungup 'IAX2/voctel-3' == Spawn extension (longdistance, 1766262, 1) exited non-zero on 'SIP/27-51de' What my IAX provider sees on the other end is this: -- Executing Dial(IAX2/tor-hub-13, Zap/G1/1766262||g) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G1/1766262 -- Zap/21-1 is proceeding passing it to IAX2/tor-hub-13 -- PROGRESS with cause code 31 received -- Zap/21-1 is making progress passing it to IAX2/tor-hub-13 -- Hungup 'Zap/21-1' I did a search through the mailing list and in the wiki. I found that cause code is used to report a normal event only when no other cause in the normal class applies. and #define AST_CAUSE_NORMAL_UNSPECIFIED 31. I am running Asterisk 1.2.0 and I am not sure what my provider is using, some version of HEAD is all I know. I am at a loss... I don't know the last time I tried to dial a toll-free from here, but it was working. Can anyone help steer me in the right direction? Thanks! Dana ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zhone Channel Bank
Has anyone successfully connected a Digium T100P to a Zhone Z-Plex 10 24 S/O? I have been unsuccessful in getting the T1 to sync up. I have searched the documentation and concluded that a cross-over cable and ESF/B8ZF configuration on both hardware should have cleared alarms but that does not seem to be the case. I would be interested in knowing the configuration and cabling on any successful installations. Thank you. Jonathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wildcard X100P/India
Can anyone tell me if they have successfully deployed the X100P in India or any where in Southeast Asia? Thank you, Jonathan ___ 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] Newbie to Asterisk - VoIP end-to-end
check out: http://www.voip-info.org/wiki-Asterisk+phones At 12:23 AM 10/15/2004 -0700, you wrote: Hi, After reading up on the Asterisk, I have a question: 1. Is there a software phone running on PC as a client that is compatible with Asterisk? My reason for asking is that I wonder if I can run voip end-to-end with Asterisk in between. Diagram: NetMeeting -- IP -- Asterisk -- (NetMeeting or its equivalent) or Cisco AS5xxx -- IP -- Asterisk -- (NetMeeting or its equivalent) Thanks, Kasey ___ 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] IP Phones ?
Not the cheapest ($75-80) but they look interesting. http://ipphone.eezeephone.com/ Jonathan At 03:10 PM 9/26/2004 -0300, you wrote: On Sun, 26 Sep 2004 19:04:38 +0100, Paul Tyreman [EMAIL PROTECTED] wrote: Hi guys, I know this isn't strictly about Asterisk, but it is related... I am looking to buy a few IP phones, but I don't have a huge budged (hence why I love Asterisk, its amazing and free !), so I was wondering if anyone knew where I could get some cheap IP Phones ? Ideally they should be no more then about £50 ($90). Thanks, Paul. The cheapest I found was the grandstream budgetone. USD 65. I'm also interested in cheap IP phones, so any news would be appreciated. Marconi. ___ 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] Digium and mailing lists
Another solution would be to keep the discussions on topic and open up a separate mailing list for people interested in open discussions. Jonathan At 07:17 PM 9/26/2004 +0100, you wrote: I was somewhat concerned reading Mark's posting earlier today. Obviously, things are very bad in the US at the moment. Their Government even deported Cat Stevens the other day (check http://news.bbc.co.uk/1/hi/england/london/3686992.stm ). Clearly, given the fact that Digium contributes so much to Asterisk, they shouldn't be forced to risk their company's future by hosting these mailing lists in such an unstable environment where they could get sued for any ridiculous reason. Even an unjustified, ambit claim could generate huge defence costs on Digium's part, and cripple their ability to contribute to Asterisk. Therefore, it seems to be in the best interests of Asterisk's `security' to have the mailing lists hosted by someone other than Digium and maybe in a country that doesn't prohibit freedom of expression. I would certainly be willing to organise hosting through another company that wouldn't be at risk from vexatious legal claims. This would allow genuinely open discussion on the lists and would mean that no messages would need to be censored from the archives. ___ 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] SMP support
I am new to Asterisk and I am investigating setting up a very large Asterisk server farm. I have found a lot of good information on this topic on the Wiki pages. I am drinking from the fire hose and I thought that I read somewhere on Wiki a caution about a potential problem with running Asterisk on an SMP system. I cannot find that info now and most of the reading seems to indicate that SMP stability is good. Does anyone know of the warning I read? Thank you. Jonathan Augenstine [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
