Re: [Asterisk-Users] NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1
Lance, I was in a similar situation, though i was rec'ing the event 6 message, i noticed no degradation of sound and so ignored it. I've since removed a *load* of unused modules, and it appears that the message is no longer coming in. I had read that some people were only getting the message after the machine had been up for a few days.. I'll check back then. This is what i added to modules.conf: noload => res_musiconhold.so noload => pbx_wilcalu.so noload => app_image.so noload => app_url.so noload => app_adsiprog.so noload => app_getcpeid.so noload => app_milliwatt.so noload => app_zapateller.so noload => app_festival.so noload => app_lookupblacklist.so noload => app_random.so noload => app_ices.so noload => app_nbscat.so noload => app_zapras.so noload => codec_adpcm.so noload => cdr_sqlite.so Lance Grover wrote: Does anyone have comment on this? I am getting: NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 on my asterisk box and it seems to be causing a poping sound in the phones, I am wondering if anyone can shed some light on this. I have scanned the archives and get possibilities ranging form motherboards, to pri, to loaded module problems. Can someone tell me the best way to start tracking this down? -- Thanks, Lance Grover ___ 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 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
Re: [Asterisk-Users] NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1
Also, around the same time, I isolated the IRQ that my zaptel cards were on. (so neither zaptel card shared its IRQ). you can see what IRQ's are in use with lspci -vb This is more likely to be the cause of the fix. Adam Dobrin wrote: Lance, I was in a similar situation, though i was rec'ing the event 6 message, i noticed no degradation of sound and so ignored it. I've since removed a *load* of unused modules, and it appears that the message is no longer coming in. I had read that some people were only getting the message after the machine had been up for a few days.. I'll check back then. This is what i added to modules.conf: noload => res_musiconhold.so noload => pbx_wilcalu.so noload => app_image.so noload => app_url.so noload => app_adsiprog.so noload => app_getcpeid.so noload => app_milliwatt.so noload => app_zapateller.so noload => app_festival.so noload => app_lookupblacklist.so noload => app_random.so noload => app_ices.so noload => app_nbscat.so noload => app_zapras.so noload => codec_adpcm.so noload => cdr_sqlite.so Lance Grover wrote: Does anyone have comment on this? I am getting: NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 on my asterisk box and it seems to be causing a poping sound in the phones, I am wondering if anyone can shed some light on this. I have scanned the archives and get possibilities ranging form motherboards, to pri, to loaded module problems. Can someone tell me the best way to start tracking this down? -- Thanks, Lance Grover ___ 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 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 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
Re: [Asterisk-Users] Voicemail
I haven't tested this, but we've been thinking the same. http://lists.digium.com/pipermail/asterisk-users/2004-November/072387.html Carlos Alperin wrote: I need to see if we can make the voicemail do the following: After they reach the voicemail, and left a message or not, they should be able to dial 0 to reach back the main menu or and operator. Any ideas? Thanks, Carlos Alperin ___ 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 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
Re: [Asterisk-Users] Definitive CallerID Format and anonymous?
Rich Adamson wrote:
Thanks for the thorough reply.
I'm aware that there necessarily are inconsistencies between termination
providers; I was just curious to find out if there's some form of
standard one should follow, which may either result in more consistent
behavior, or at least shift culpability to the provider...
Bottom line is that I want to tell my IAX providers "Hey, I'm doing the
right thing, could you find out where it's breaking?"
Here's what I've been told to use and it works with teliax & Nufone:
exten => _1NX,1,SetCallerID(4025551212|a)
exten => _1NX,2,SetCIDName(NPI|a)
exten => _1NX,3,Dial(IAX2/teliaxout/${EXTEN})
The setcidname isn't really needed under most circumstances, but I
send it anyway for consistency.
I've been messing around with mine, and found the same. It seems the
CIDName is looked up by the rec'ing party; so SetCIDName does nothing.
Also maybe of note, when i try to set the CIDNum ANI (|a) its getting
jacked back to my primary DID (whereas without the |a it changes the
CIDNum).
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[Asterisk-Users] Zaptel Fax Detection
I've read that the auto fax detection for asterisk is built into the chan_zap software, however i've been experiencing odd behavior. I have two digium cards, a TDM and a TE110p, and the only time i am getting the fax detection is on outgoing calls from the TDM card. zapata.conf has =both. the TE is a PRI interface to a T1 with 8 channels for voice. ___ 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
Re: [Asterisk-Users] Polycom Auto-Answer problems
Chad Osmond wrote:
CVS Head from 07/07/2005
I'm trying to make an IP-501 auto answer a call.
exten => 301,1,SetVar(_ALERT_INFO="Ring_Ans")
exten => 301,2,SetVar(ALERT_INFO="Ring_Ans") # Tried both combinations
exten => 301,3,Dial(SIP/5001,15)
exten => 301,4,Hangup
Sip.cfg for Polycom phone
Ipmid.cfg
you um, rebooted the phone, right? (and are sure the new configuration
was loaded?)
Asterisk Log:
-- Executing SetVar("SIP/5002-6e20", "_ALERT_INFO="Ring_Ans"") in new
stack
-- Executing SetVar("SIP/5002-6e20", "ALERT_INFO="Ring_Ans"") in new
stack
-- Executing Dial("SIP/5002-6e20", "SIP/5001|15") in new stack
-- Called 5001
-- SIP/5001-f735 is ringing
-- Nobody picked up in 15000 ms
As you can see it just rings, and then hangs up.
Any one have an idea?
Chad
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Re: [Asterisk-Users] initiate call with asterisk
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Eric wrote: I would like to initiate a call in asterisk (say with cron) so that this call rings on the destination number _and_ on an asterisk extension. How would I achieve this? thanks ___ 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] Call quality degradation after time
I'm using Polycom 501's; with stable1.0.8, g729 and a very decent machine; we have a PRI interface to a T1. Many users complain that after a given amount of time, say, 30 or 40 minutes on a call, the outside party complains that their sound keeps 'cutting in and out'. I believe that the incoming sound quality remains fine. I had read that there may be some soft of memory issue with the Polycom's... what else could be causing this? Will auto-rebooting the phones once every few days fix this problem? Thanks. ___ 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
Re: [Asterisk-Users] RE: Business Edition
here, here! Kevin P. Fleming wrote: Lee Howard wrote: Go ahead and have a proprietary fork, sell it, have it specially licensed. But please, please, please treat the community fairly. Otherwise it causes unrest in the community, discourages contribution, encourages forking, and triggers forum threads like this one. You seem to be neglecting the amount of work that Digium puts into the Asterisk (and related) products on an ongoing basis that is given to the community at no charge. Saying "Digium GPLed Asterisk and we're thankful for that, now what" only makes sense if it happened one time and then Digium stopped contributing to the source base. I think the situation is much different from that (although I'm obviously biased since I'm paid to work full-time on Asterisk by Digium). Digium continues to pay people to provide enhancements to Asterisk, Zaptel and the related projects, and those enhancements are even used by companies that directly compete with Digium. We also very strongly push our "custom development" customers in the direction of letting us include the code we write for them into the open-source tree, so that the community will benefit. In other words, I think the contributions from Digium to the community are ongoing, and in many ways more than offset the license that contributors grant to us for the commercial use of their code in our products. Your opinion may vary, of course :-) ___ 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 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
Re: [Asterisk-Users] caller id on a T1 PRI
"pri debug span 1" will show you what the PRI is sending to you, before you do that: have you tried the previous suggestion of issuing a Wait(1) (i actually suggest 2) prior to Answer or Dial? Ryan Williams wrote: How do I go about doing this? I can not find any documentation on "isdn trace" Should I be receiving CID Name as well as number if I am on a PRI line? I am fairly certain it was working when we had POTS lines. -Ryan -Original Message- From: Paul Belanger [mailto:[EMAIL PROTECTED] Sent: Thursday, July 21, 2005 6:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] caller id on a T1 PRI Ryan Williams wrote: I understand how CID works and how you must set CID when dialing out on a PRI and how the phone company sets the name. I was wondering how this works in regards to inbound calls. I have a pri and I get the number that the caller is coming from but I do not get the name. Is this normal? How do I end up getting the name on this PRI as well? It could be possible the telco is not sending it to you. You could always setup an ISDN trace and post the results; just to be sure. PB ___ 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 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 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
Re: [Asterisk-Users] Call quality degradation after time
Thanks for the reply, Adam. If this is the case, it would seem to me (because the degradation happens only after a period of time, and quite suddenly) that the issue lies with digium's implementation of g729. As an interesting note, I had the same problems using ulaw -> ulaw over the local network (from internal phone to internal phone) with a much shorter period of 'good voice' before degradation--which was my reason for switching to g729, which seemed to solve the problem. If there is no other current solution, it would seem to me the best thing to do would be to force g729 for non PSTN connections, and ulaw (im in the US) for calls going out the PRI. Has anyone done this..? I'm still hoping to be able to stick with g729; anyone else experience this kind of issue? -a Adam Goryachev wrote: On Thu, 2005-07-21 at 15:56 -0400, Adam Dobrin wrote: I'm using Polycom 501's; with stable1.0.8, g729 and a very decent machine; we have a PRI interface to a T1. Many users complain that after a given amount of time, say, 30 or 40 minutes on a call, the outside party complains that their sound keeps 'cutting in and out'. I believe that the incoming sound quality remains fine. I had read that there may be some soft of memory issue with the Polycom's... what else could be causing this? Will auto-rebooting the phones once every few days fix this problem? We had this problem initially, except it happened from the beginning of the call. The fix was to tell the phones to prefer alaw and to tell asterisk to prefer alaw (and not allow anything else). This meant that we were using alaw from PSTN -> asterisk -> SIP phone ie, everywhere. It solved the above symptom. PS, this is in Australia and was with CVS stable 1.0.x from about 12 months ago. Regards, Adam ___ 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 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] CVS-HEAD dies signal 11 after incorrect vm password
anyone else have the above issue? this is today's CVS. thanks. ___ 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
Re: [Asterisk-Users] Opteron Hardware with Asterisk
I have asterisk running on dual 244's. Everything works fine, the only special issue i had was installing the g729a codec (required a very tiny tweak to the asterisk Maiefile). Unfortunately, the system doesn't get a huge amount of traffic, so I can't testify to capacity. Running 1.0.8, btw. Asterisk Supporter wrote: Anyone running Asterisk on dual Opteron Server? Are there any special issues in a 64 bit environment and what is the capacity curve like? ___ 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 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
Re: [Asterisk-Users] Opteron Hardware with Asterisk
Sarge. RHEL/compatible should be fine too. Wiley Siler wrote: Did you build it using the 64 bit CentOS or another Distro? Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Dobrin Sent: Friday, July 22, 2005 4:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Opteron Hardware with Asterisk I have asterisk running on dual 244's. Everything works fine, the only special issue i had was installing the g729a codec (required a very tiny tweak to the asterisk Maiefile). Unfortunately, the system doesn't get a huge amount of traffic, so I can't testify to capacity. Running 1.0.8, btw. Asterisk Supporter wrote: Anyone running Asterisk on dual Opteron Server? Are there any special issues in a 64 bit environment and what is the capacity curve like? ___ 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 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 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 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
Re: [Asterisk-Users] Mixed Voice/Data T1
As nice as HDLC sounds in theory; we have the same setup, a T1 with afew lines split off, and i just don't see a need to add the routing load to the asterisk machine. We have an Adtran 604 which splits the T into a PRI and 10/100. Incidentally, HDLC in asterisk seems to be.. a hassle to get working at best. William Boehlke wrote: The voice side of a fractional T1 is usually delivered as analog connections. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Wednesday, July 13, 2005 12:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Mixed Voice/Data T1 We have a server running Asterisk and shorewall, three network interfaces and a T1 card, it functions as our firewall, pbx and connects to an Adtran 600 for FXS/FXO. We currently have two internet feeds, hence the three NICs. Our 10 PSTN lines are currently delivered POTS to our FXO and rather lousy service, volume is low and voice quality is dull. The Telco is offering me a Fractional T1 for our data needs. I thought that the best way would be to have the voice and data delivered on the T1. How difficult would it be to take the T1 into our Sangoma T1 card and seperate out the voice to asterisk and the data as our internet feed? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.14/48 - Release Date: 7/13/2005 ___ 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
Re: [Asterisk-Users] Call quality degradation after time
Thanks for the reply, Adam. If this is the case, it would seem to me (because the degradation happens only after a period of time, and quite suddenly) that the issue lies with digium's implementation of g729. As an interesting note, I had the same problems using ulaw -> ulaw over the local network (from internal phone to internal phone) with a much shorter period of 'good voice' before degradation--which was my reason for switching to g729, which seemed to solve the problem. If there is no other current solution, it would seem to me the best thing to do would be to force g729 for non PSTN connections, and ulaw (im in the US) for calls going out the PRI. Has anyone done this..? I'm still hoping to be able to stick with g729; anyone else experience this kind of issue? -a Adam Goryachev wrote: On Thu, 2005-07-21 at 15:56 -0400, Adam Dobrin wrote: I'm using Polycom 501's; with stable1.0.8, g729 and a very decent machine; we have a PRI interface to a T1. Many users complain that after a given amount of time, say, 30 or 40 minutes on a call, the outside party complains that their sound keeps 'cutting in and out'. I believe that the incoming sound quality remains fine. I had read that there may be some soft of memory issue with the Polycom's... what else could be causing this? Will auto-rebooting the phones once every few days fix this problem? We had this problem initially, except it happened from the beginning of the call. The fix was to tell the phones to prefer alaw and to tell asterisk to prefer alaw (and not allow anything else). This meant that we were using alaw from PSTN -> asterisk -> SIP phone ie, everywhere. It solved the above symptom. PS, this is in Australia and was with CVS stable 1.0.x from about 12 months ago. Regards, Adam ___ 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 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 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
Re: [Asterisk-Users] Can you caculate with me?
Bob Goddard wrote: On Thursday 28 Jul 2005 13:07, Ronald Wiplinger wrote: before I accuse somebody to "overbill" I would like you to calculate with me: Rate: 0.0189 for calling Taiwan via NuFone Duration: 930 seconds Lets vote for the answers:0.7269 or 0.2929 ??? Assuming it is per minute; 930 * 0.0189 / 60 = 0.29295 B I get .31$. Where did you all go to school? Is there a connection charge? ___ 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
Re: [Asterisk-Users] delay on pri dialling when asterisk is spliced between E1-Pri and legacy pbx
You aren't dealing with analog phones, and you aren't transmitting DTMF signals.. the functional difference between analog and digital systems kindof precludes what you are looking to do.. meanwhile, once the entire number has been dialed, the outgoing call should be started almost instantaneously.. maybe set the initial context so that the longest dial string is the length of the extension..? Frank Sautter wrote: Maik Schmitt schrieb: one of our customers which wants a soft transfer between his old pbx to asterisk and sip. the setup is as follows: telco <---pri---> asterisk <---pri---> legacy pbx everything is fine exept that when dialling from the legacy pbx it takes about 3 seconds before the asterisk start to dial. where does this delay come from? has it to do with 'overlapdial=yes'? This is normal behaviour if you use '.' in your extensions.conf. Use '!' instead and Asterisk will start dialing immediately. when i change '.' to '!' then the overlap digits get lost. this means the longest number dialled on my telco line is as long as there are abigous matches in the dialplan. isn't there a way to start dialling after one received enough digits to decide which path to dial and then still transmit the remaining (overlapping) digits? regards frank ___ 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 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
Re: [Asterisk-Users] Minimum CPU required for 60 calls
And as much as you dislike these kinds of questions; its unfortunate that the community doesn't have any good answers to them available--they should be. It would be great if we could get some independent verification of digium's claims/figures. voip-info: http://www.voip-info.org/tiki-index.php?page=Asterisk+setup+medium+office+100 Andrew Kohlsmith wrote: On Tuesday 02 August 2005 06:16, Obelix wrote: I am interested in how much CPU and RAM asterisk requires for call handling. I *really* dislike these kinds of questions. Grab some hardware and try it. It is the *ONLY* way you will know for sure. Grab a single processor Pentium 4 or Celeron system and do some testing. I'm sure you have one sitting around somewhere you can use for a test, even if you have to put a different hard drive in it for the test. Typically speaking, if you have to ask these kinds of questions you are NOWHERE near the level of competence in Asterisk to try and skimp and save on the hardware. That is not meant as an insult, either. Get it working, THEN start looking to pinch the pennies. You will only be disappointed otherwise. -A. ___ 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 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
Re: [Asterisk-Users] Polycom FW
www.emplifyhr.com/pcom Douglas Garstang wrote: We purchased our phones through Alliance Systems, a Polycom certified reseller. Getting firmware was difficult, and they where very unresponsive, probably because we didn't pay them additional money for a support contract. Such is life. -Original Message- From: The VoIP Connection [mailto:[EMAIL PROTECTED]] Sent: Thursday, January 19, 2006 8:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom FW Bill, If you purchased your phones from a certified reseller they should be able to get you these files. If you didn't, contact me off-list. I can help you. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Bill Michaelson [mailto:[EMAIL PROTECTED]] Sent: Thursday, January 19, 2006 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom FW Anyone know how to obtain firmware and starter .cfg files for Polycom phones? Despite registering at the Polycom web site, I can't locate this stuff. ___ --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
