Re: [asterisk-users] Unprovisioned 7961
Try this configuration file... http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configu ration+files+for+SIP#Cisco7961with833SR2ConfigurationExamples Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Wong Sent: Friday, January 25, 2008 6:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Unprovisioned 7961 Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says Error Verifying Config Info. I have read quite a bit on this topic (getting 7961's to work with Asterisk and TB) and only came across a few postings where other people encountered this issue but no solution was given. I have checked the SEP.cnf.xml file for the phone and everything seems to be right. I even tried to remove some parts of the code as people suggested but no luck. I already have a 7960 on TB so I know that TFTP is working correctly. Any ideas on how I can get this to work would be much appreciated. Thank. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] Cisco 7941G-GE with Asterisk and CTPSEP odyssee
I've always found it best to run with no CTLSEPMAC.tlv file in the tftp server directory; it will ignore that and move on. With the 7961's you'll be best in the 8-3-3SR2 leading edge, the DND button is where it should be, and transfers work the way they should again. The XML configuration file is different, I'll post one onto the Wiki that actually works and takes updates. Can you post a pinout of the cable that you're using for serial? I have a 6 Wire cable and a RG12 (6 Wire) to DB-9, but no working pinouts... Ideally from End to End pinout would be great. Thanks, Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: January 4, 2008 10:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee Christophorus Laube wrote: Hi list, I have bought some Cisco 7941G-GE IP phones and want to use them with asterisk. Before bying I tested the whole setup with three different models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the formerly provided SCCP-Image to SIP was no problem, but now it complains about a nonexistent CTLSEPmac.tlv file. Most of the howtos say something about an empty file but that does not suit to me. Don't know what you mean by that does not suit me Simply create a file with no contents and the load will be happy There is plenty of information available on how to up the 7960 from SCCP to SIP, and I assume from your question you are in the transition from 6 to 7. Use Google, look on the voip-info site and you will find step by step procedures. Different versions of now obsolete firmware are also on the Net. 7961, 7970 results may vary! Does anyone of you have experience in getting these phones to work or can point me to any information bringing me back in the game? You need to work your way up, version by version as well. The problem I now have is version 7.1 works fine, version 7.3 won't place any calls and goes to reorder. Asterisk shows it trying to go to a non existent context, not specified in sip.conf Any ideas? __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ --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] Cisco 7961 new firmware stops readingconfiguration files
I contacted one of the list users and they sent me their configuration files. I used it as a template and it worked with my phone, so I'll be sure to put it back up on the Wiki. Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Preston Edwards Sent: December 20, 2007 6:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7961 new firmware stops readingconfiguration files Chad, I had the same problem when upgrading to some of the newer firmware. The newer firmware gets even pickier (if that's even possible) about the config files. Go the phone's webpage and look at the debug log. It will show you where it's not parsing correctly. I'm not in front of my phone now so I can't look, but I remember it getting upset about networkLocale or userLocale or something of that nature, so I just removed that section of the XML code and it loaded fine. Good luck, Preston Edwards __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email _ --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] Cisco 7961 new firmware stops reading configuration files
Hello, I have been running SIP41.8-2-2SR4S firmware on the Cisco 7961 and have recently upgraded to SIP41.8-3-2SR1S to add the DND button on the front of the phone and also to hopefully resolve some issues with the phones not registering after a long period. Once we upgraded the phones now display Error Verifying Config Info in the Status messages and will not process the configuration file. To make a change on the phone I have to downgrade to 8.2.2R4 and change the configuration, and then upgrade to 8.3.2R1, which is a bit of a pain. The tftp logs indicate that the phones is getting the correct SEPMAC.xml.cnf file but will not parse it, so that doesn't seem to be the issue. The Wiki pages for 79x1 indicate that it's a known issue, has anyone managed to get past the issue? I tried logging a call with Cisco TAC, but they're giving the We don't support SIP on anything other then CME... Thanks, Chad __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ --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] Sangoma Dell 750
The A101 does not require a PCI-X. I have 2 A102's running in standard ports here. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duracom Lists Sent: November 27, 2006 2:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Sangoma Dell 750 Anyone using a Sangoma A102 with a Dell 750? We are looking at going this route but needed some input. I really only need a Single T1 port, but this server doesn't have a PCI-X port, which the A101 apparently requires? Thoughts, Suggestions? K ___ --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] T1 Copper or T1 Fiber Line
T1 PRI's are (almost?) always copper. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: June 16, 2006 5:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] T1 Copper or T1 Fiber Line I am located in the Caribbean, we already have an Asterisk PBX box, and are planning on getting a T1 line. The Phone company said they will run the T1 line over copper. So basically it's a T1 copper line. Is something like that possible? and if so, what would I need to get and do to be able to get this type of T1 line to work. Thanks Dakota ___ --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] Compiling zaptel on FC5
Try "rpm -qa kernel-devel" or "rpm -qa kernel-smp-devel" to verify they're installed. If not do a "yum install kernel-devel or kernel-smp-devel" depending on which you have. Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J.J. FeminellaSent: June 13, 2006 9:51 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Compiling zaptel on FC5 Are there any generic install guidelines for compiling the Zaptel drivers on FC5? This is my first install of Asterisk (and my first FC5 system) and I'm having a great deal of trouble getting it to cooperate. make clean and make are definitely not playing nice, telling me that "You don't appear to have the kernel sources installed" when I'm pretty sure that I do. Any pointers? thanks, JJ ___ --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] Polycom - missed calls dial back
Prepend outgoing calls by using Dial(9{exten}) instead of dialing 9 to
get out on your system...
Or, add a 9 to caller id.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
GibbsSent: May 18, 2006 11:37 AMTo: Asterisk Users Mailing
List - Non-Commercial DiscussionSubject: [Asterisk-Users] Polycom -
missed calls dial back
This is not necessarily Asterisk
specific but if I have Polycom 301/501 and 601s and want to dial a missed call
back, how do I prepend a 9 can I do this via the polycom config? I cant
find anything in the docs.
Bill
___
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RE: [Asterisk-Users] Tool for Polycom configurations
You can use my script, based on Chris Mason's script, to do most of what you want, you can feed it your MAC's and Extensions and it will create the phones. Be warned, it's not pretty, my perl book was in storage so I did a lot of kludging. Feel fee to update. http://holburn.com/poly/poly-add-phone.pl Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce ReevesSent: May 4, 2006 2:45 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Tool for Polycom configurations I am getting read to roll out close to 100 polycom phones and wondered if any one knows of a program to take a list of MAC addresses, extensions, and names and generate the configuration files? ___ --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] Polycom NTP issue
Are you passing the Offset through the DHCP server as well? On a linux DHCP server this would be: option time-offset -18000; # Eastern Standard Time option ntp-servers 192.168.x.x The fact that the date is wrong, but the time is correct, seems a little strange to me. Are you sure your Windows server is set to the correct date? What happens if you set the time on the phone to a different time and date, will it pull the date back of the NTP server? Also, try and run some network sniffing tools to verify that it is getting a valid response. I am ready to pull my hair out. I cannot seem to get the Polycoms to read the time properly. Regardless of the server they are pointed to our the offset, i am getting the correct time, but 24 hours ahead. So for today it is showing Friday April 28 but with the correct time. Any clues? ___ --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] Looking for input on which way to go with smallbusiness setup
If you use a Sangoma T1 card, (A10x) card you can send both voice and data down the same T1 and have the Sangoma card split it for you. If you are talking about non-hobby usage, stay away from FXO adapters and go with a T1.. You'll be much happier in the long run. For a fractional T1, don't worry about hardware echo, asterisk will handle the echo fine. I would seriously consider shelling out for a decent phone. For example, the Polycom 301 is an entry level phone that isn't too expensive. Your users will hate using an analog phone for transferring and forwarding. If you decide to run analog phones, then you may as well get a multiport FXS to T1 device and then buy a 2x T1 card like the A102 from Sangoma, or the Digium equivalent. I would suggest that the 768k/s fractional T1 will cost you more then a 3Mb/512Kb DSL line, so go with the DSL if your SLA can live with the occasional outage. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T. Shaw Sent: April 27, 2006 4:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Looking for input on which way to go with smallbusiness setup Hey guys! I'm the past week and a half, I have really learned a lot from the mailing list and the wiki's posted online. Now I have a question regarding different ways I can setup my asterisk server for a small business with 12 extensions in the office. Cost is a great concern, so I know cheap analog phones at the desks is what we are looking at. My question is, should I go do a fractional T1 for voice only, and the get a Internet Service provider for their data needs, get some sort of ATA and run the 12 analog phones to that OR go with what SBC/ATT (in my area) is offering, and do the Integrated T1 option and have just the 12 channels for voice, run some ATA between * and the phones and use the 768 up and down for data? Anyone use this Integrated T1 with asterisk before? What hardware did you use? Thanks for your input! Terrelle ___ --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] Power over Ethernet (PoE) switch recommendations
SMC 6824MPE.. Does 24 ports POE, with 2x 1GB uplinks, (RJ45 or GBIC) L3 Managed switch. We've got four of them here, and I think they're great, the cost was really reasonable. Ingram no longer lists the MPE model, but it should be available still. Chad -Original Message- Hi listers, I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's. I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office. However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in price. I would appreciate any input people have to offer. ___ --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] Callid on T-1 trunk
Yes, It can send Caller ID down the T1 line.. Some T1's only accept Caller ID's that match the set of DID's associated with the T1. Others, like mine, will take anything you send it... Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jordan NovakSent: March 30, 2006 8:18 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Callid on T-1 trunk I am not getting any caller Id with my standard T-1. Is a standard T capable of sending callerid? I dont want to spend time troubleshooting my PBX if Asterisk cant send it down that type of trunk. Jordan ___ --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] Polycom 501 power over ethernet
I was just thinking, about this.. Move your Polycom Power Injecting Patch cable (Black Cable with AC Adapter Input) into the cabling closet. You could then infuse the power at the cabling closet and then just use a standard patch cable to patch the phone in. You would be looking at a line loss of 40 Ohms per 1000 ft, or about 12 Ohms per 300ft run. Max output of the transformer is 400mA @ 12V The Voltage drop of a 12 Ohm load on a 400mA circuit is 0.03V... So that should be more the acceptable. I just don't know what would happen if a user plugged a phone into the line.. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William M Conlon Sent: March 5, 2006 8:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet My recollection of the marketing fluff was that we would just use our legacy network (cables) and the devices at both ends would figure out whether they were sourcing, sinking, or neither. In the case of the 501, it's the special Polycom cable, either with or without provision for an AC power adapter, that powers the phone. That's what I meant by saying the '501' itself is not compliant with 802.3af -- it needs a separate thingamajig [tech jargon :)]to be powered. Anyway I had hoped that I could just plug a CAT-5 patch cable from my RJ45 wall outlet into the phone. On Mar 5, 2006, at 5:17 PM, Michael Welter wrote: As I understand 802.3af, the phones go through a negotiation with the unit supplying the power. I don't think it's a matter of -48VDC on a particular pair. I remember a schematic from years ago--it had each of the receive pair and the transmit pair going into a transformer winding, and that winding had a center tap for PoE. This is not something that *I* am going to screw with. The IP501 telephone set is the same for both PoE and local power. With the PoE cable, the 802.3af electronics (the negotiator) is a plastic thing in the cable. For the local power, there is a plastic thingie toward the wall end of the cable, and you plug the wall wart into the plastic thingie. Notice the advanced technical jargon here With local power, there is still only one cable one the desk--the power plugs into the cable towards the wall. Except for a power interruption, this has all the advantages of PoE. William M Conlon wrote: I saw that Polycom offered a cable (not stocked anywhere), at $40 a pop for 802.3af connections. That's what made me think the phone itself is NOT 802.3af compliant. Presumably, for $40, there's more than a fuse in that special cable. On Mar 5, 2006, at 4:31 PM, Paul Hales wrote: For Polycom IP500/501's and IP300/301's you need a special polycom POE cable. When you buy Polycom phones you can usually specify POE or powerpack. PaulH On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote: When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were powered over ethernet. Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear [EMAIL PROTECTED]) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power injector cable, they work with the switch in either its powered or unpowered ports. Anyhow, I hadn't seen any mention of how people power these phones, as I had planned on centralizing phone power on a UPS to supply my Asterisk server and POE switch. Now the question is: Can the Polycom AC-powered injector be used with a standard ethernet patch cable: switch :: Polycom injector cable :: RJ45 coupler :: patch cable :: Polycom 501 which would allow me to power the Polycom AC adapters by my UPS. Or do I need to provide a UPS at each phone and run the ethernet like switch :: patch cable :: RJ45 coupler :: Polycom injector cable :: Polycom 501 thanks. ___ --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 Bill William M. Conlon, P.E., Ph.D. To the Point 345 California Avenue Suite 2 Palo Alto, CA 94306 vox: 650.327.2175 (direct) fax: 650.329.8335 mobile: 650.906.9929 e-mail: mailto:[EMAIL PROTECTED] web: http://www.tothept.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
RE: [Asterisk-Users] co-location providers in Ottawa, Canada
Telecom Ottawa? Large, Ultra fast pipe with direct connections to TDM providers (Which may be at 151 Front St. in Toronto) but they should work for what you want. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard OSSSent: February 19, 2006 12:04 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] co-location providers in Ottawa, Canada Anybody know ifthere are co-location providers in Ottawa, Canada? We are planning on co-locating our Asterisk conferencing server. One more thing, is there an interest in reviving the Ottawa Asterisk User Group? Seems like the original group has been inactive for quite awhile. I will volunteer to organize it. Thanks. richard ___ --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 vs. Traditional PBX
Title: Asterisk vs. Traditional PBX You may be using less then ideal phones. With a Polycom 501, I can't see you having voice quality issues, With a Sangoma or Digium card and a PRI the quality and functions of a Asterisk system are on par with most PBX's (I'd say they're above). It is a good solution for most companies, consider the ability to change features and expand only limited by your abilities (or those of consultants). For 200 people, you will probably need40 channels, which will be two T1's, so start looking for a dual T1 card ( again Digium and Sangoma make excellent products). Hope this helps, there are thousands of systems running in companies of your size. I would recommend running two servers in a active/passive format and rsync them every hour (to a different directory). If the server blows up and kills the board you can easily switch over in a few seconds. It also makes upgrading easier,. Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nora LavelleSent: February 9, 2006 9:15 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk vs. Traditional PBX Hi everyone !So here's my question of the day ! I need to make a decision on whether or not to go to a voip solution or configure an existing pbx (norstar) that my company has available. We are a small startup. I'm wanting a solution that will support up to about 200 people, with direct dial-in capability, up to about 30 concurrent phone calls and good voice quality. Right now I have an asterisk deployment with about 15 people on it. We have sipura 841 phones. The biggest issue currently is voice quality. lot of complaints there. I have a dell 650 poweredge (single processory system), with a digium tdm400 card and 4 analog lines plugged into it.So here are my questions:* Is asterisk a good solution for my company ? or should I just install the traditional pbx and look to move to asterisk in a couple of years ? (I personally would prefer asterisk cuz I'm a unix person not a phone person so from a manageability perspective i would love this )* If I were to go to an asterisk solution to support about 200 people with the requirements above what hardware platform would you recommend ? I'm guessing I'd need a PRI line and a different digium card? Also would a 1cpu poweredge dell be enough ? or would that have to be upgraded too ?If anyone is running an environment similar to this that can provide help I would really appreciate this. I'm having a hard time making this decision and would love to hear anybody's experience in a real time environment.Thanks again this list ROCKS!Nora Lavelle ___ --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] Caller stuck in MoH after being answered by a phone that was forwarded to.
Can anyone shed some light on what happened? Asterisk 1.2.1 with Zaptel 1.2.1 Here is what I know happened: A call came into our main number and was answered Asterisk set the monitor CALLFILENAME and then started monitor. The call was directed to a context called open where all calls go during business hours. The dial plan has a Answer() again, and then played a message (custom/1) The next dial plan was for a Dial SIP/221SIP/222 statement to dial our reception phones. One of the Receptions phones was forwarded because they were out on lunch. Extension 222 was forwarded to 249 who answered the call (Polycom 501) 249 Answered the call and then transferred to another users phone (223). The phone (223) rang once and then stopped ringing. The user on Zap/1-1 was stuck in MOH until he hung up. 4:45 [21637] : -- Accepting call from '416497' to '1484' on channel 0/1, span 1 4:45 [23545] : -- Executing Answer(Zap/1-1, ) in new stack 4:45 [23545] : -- Executing Set(Zap/1-1, CALLFILENAME=i416497-20060201-143445) in new stack 4:45 [23545] : -- Executing Monitor(Zap/1-1, wav|i416497-20060201-143445|m) in new stack 4:45 [23545] : -- Executing Wait(Zap/1-1, 1) in new stack 4:46 [23545] : -- Executing NoOp(Zap/1-1, 416497) in new stack 4:46 [23545] : -- Executing GotoIfTime(Zap/1-1, 8:30-16:30|mon-fri|*|*?open|s|1) in new stack 4:46 [23545] : -- Goto (open,s,1) 4:46 [23545] : -- Executing Answer(Zap/1-1, ) in new stack 4:46 [23545] : -- Executing BackGround(Zap/1-1, custom/1) in new stack 4:46 [23545] : -- Playing 'custom/1' (language 'en') 4:53 [23545] : -- Executing Dial(Zap/1-1, SIP/221SIP/222|30|t) in new stack 4:53 [23545] : -- Called 221 4:53 [23545] : -- Called 222 4:53 [21640] : -- Got SIP response 302 Moved Temporarily back from 192.168.129.131 4:53 [23545] : -- Now forwarding Zap/1-1 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/222-e1ef) 4:53 [23548] : -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/249|30|t) in new stack 4:53 [23548] : -- Called 249 4:53 [23545] : -- SIP/221-fa0f is ringing 4:53 [23548] : -- SIP/249-579c is ringing 4:53 [23545] : -- Local/[EMAIL PROTECTED],1 is ringing 5:01 [23548] : -- SIP/249-579c answered Local/[EMAIL PROTECTED],2 5:01 [23545] : -- Local/[EMAIL PROTECTED],1 stopped sounds 5:01 [23545] : -- Local/[EMAIL PROTECTED],1 answered Zap/1-1 5:01 [23545] : == Spawn extension (open, s, 3) exited non-zero on 'Local/[EMAIL PROTECTED],2ZOMBIE' 5:17 [21640] : -- Started music on hold, class 'default', on channel 'Zap/1-1' 5:21 [23564] : -- Executing Dial(SIP/249-a869, SIP/223|30|t) in new stack 5:21 [23564] : -- Called 223 5:22 [23564] : -- SIP/223-885d is ringing 5:24 [23564] : == Spawn extension (from_sip, 223, 1) exited non-zero on 'SIP/249-a869' 8:33 [21637] : -- Channel 0/1, span 1 got hangup request 8:33 [23548] : -- Stopped music on hold on Zap/1-1 8:33 [23548] : == Spawn extension (from_sip, 249, 1) exited non-zero on 'Zap/1-1' 8:33 [23548] : -- Hungup 'Zap/1-1' Chad ___ --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: Using *RT for HA purposes was: [Asterisk-Users]RealtimeMultipleAsterisk boxes, iaxusers
Realtime.. As in pulling configs from a realtime database.. Or he's trying to link Asterisk to www.bestpracticals.com version of Request Tracker (also known as RT) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Charles Wang Sent: February 3, 2006 8:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Using *RT for HA purposes was: [Asterisk-Users]RealtimeMultipleAsterisk boxes, iaxusers Hi, ALL: Can anyone tell me what *RT is ? What is its full name? I think the * is asterisk but what is RT ? 2006/2/2, Rusty Shackleford [EMAIL PROTECTED]: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alistair Cunningham Sent: Wednesday, January 04, 2006 4:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Using *RT for HA purposes was: [Asterisk-Users]Realtime MultipleAsterisk boxes, iaxusers load balacing isn't perfect, and it can give uneven loads at low capacity, but it gets better as load increases which is where it matters. What kind of loads are we talking about here, please? ___ --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 -- Best Regards Charles ___ --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] Polycom 501 horrible echo
BootBlock 2.5.0 Bootrom 2.6.2.0032 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Senykoff Sent: January 27, 2006 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom 501 horrible echo I've been running 1.6.4.0064 for the last few weeks.. I've had no problems with it, I haven't done a whole lot of speaker phone with it yet though.. Once my IP4000 reboots It'll be running it as well so that will be a good test. Which bootrom version are you using? -Ron ___ --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] Polycom 501 horrible echo
I've been running 1.6.4.0064 for the last few weeks.. I've had no problems with it, I haven't done a whole lot of speaker phone with it yet though.. Once my IP4000 reboots It'll be running it as well so that will be a good test. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff Herring Sent: January 26, 2006 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom 501 horrible echo Now I'm really confused... 1.6.3 is on the Polycom Website as the latest... I'm running 1.6.2.0041 according to my phone. Which firmware worked for you? At 04:04 PM 1/26/2006, Ron Senykoff wrote: We also have noticed a poor server config can cause this in testing. Noticed when I had one person building * servers using Debian. Had them rebuilt with FC4 and have no issues - yet:) I recently upgraded all our phones to the latest Polycom firmware 1.6.2 and went from great speakerphone to tons of feedback. I would hate to have to go back to the old firmware. Although Polycom recommends keeping the older bootrom unless you need https provisioning, I'm going to try the new bootrom and see if it fixes the problem. This is being experienced across 3 corporate offices with 3 separate Asterisk servers. And I have to reiterate... all was good until the firmware upgrade. -Ron ___ --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] Re: Polycom 501 horrible echo
I have had no problems running the Sip.cfg from 1.5.2 with 1.6.4 so far, but I am looking to update in the next while. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: January 27, 2006 1:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: Polycom 501 horrible echo Hi - I'm running 1.6.2.0041 according to my phone. Which firmware worked for you? It was the old firmware from when we first got the phones actually. 1.4.x I think. Then I read that they fixed the CID issue and decided we needed an upgrade. I tried it out on my phone, but didn't really notice the problem until we had upgraded the rest. Oh well... Also, these are IP 500 SIP. We've been using Polycom phones since firmware version 1.3.0, and I've used every version of the firmware since then in production on IP300s, IP500s, IP501s, IP600s, IP601s and IP4000s. I've never had this issue on any of them. I don't mean to downplay the issue, but it may be possible that you did, in fact, get a bad batch of phones. When I've ordered these phones in quantity before, I've gotten many phones with consecutive serial/mac addresses, so they were probably manufactured in a bunch. Maybe a bad batch of mics got installed on a group of phones? One thing I was pondering: you are not, by chance, using the same sip.cfg between version 1.4.1 and version 1.6.2 are you? The file has changed significantly between these versions, and certain acoustic settings that worked with 1.4.1 may not work with 1.6.2 (Not to mention that ipmid.cfg and sip.cfg were merged in the 1.5.x release). - Noah ___ --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] Announcing PodMail 1.0 (GPL)
Supporting authentication directly against voicemail.conf or using an LDAP directory, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: January 23, 2006 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Announcing PodMail 1.0 (GPL) Cute? But it can use LDAP... PaulH - Original Message - From: Ben Klang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 24, 2006 3:58 AM Subject: [Asterisk-Users] Announcing PodMail 1.0 (GPL) Hello Asterisk Community. While sitting at lunch the other day I had a typical napkin-prototype idea: What if I could make my Asterisk Voicemail accessible as a Podcast in iTunes? Three hours later with the help of two friends I had a working proof of concept. Now we are releasing the polished version of this idea as PodMail 1.0 PodMail brings together open-source telephony and Podcasting to create a new, useful way of accessing voicemail and podcasting. PodMail integrates with Asterisk to provide a secure podcast of your voicemail. Supporting authentication directly against voicemail.conf or using an LDAP directory, PodMail allows you to subscribe to your own voicemail box. Each time you dock your iPod, your new voicemails will sync right along. Listen to your voicemail at your convenience and without using cell minutes. PodMail also allows for a brand new type of PodCasting. Unchain Podcasting from the computer! Configure PodMail for public access and you have a ready-to-run PodCast. Updating your Podcast is as easy as phone call. Moblogging has never been so easy or flexible. Live Demo: Do not miss out our live demo at http://podmail.alkaloid.net/ Leave us a message in one of our mailboxes, subscribe to one of the PodMail Podcasts, then see and hear your message immediately! Check out the PodMail Documentation and Installation Notes at http://projects.alkaloid.net. PodMail is released under the terms of the GPL. Enjoy! /BAK/ -- Ben Klang Alkaloid Networks http://projects.alkaloid.net [EMAIL PROTECTED] 404.475.4850 ___ --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] Slightly OT: Plantronics headset quick connectorwiring
RJ11Plantronics 13 24 31 42 RJ11 Pin 1 is on the left when looking at the contact points. Plantronics Pin 1 is on the left when looking at the contacts (through the plastic sheild) My multimeter battery is low, so YMMV, but: Pin's 1,4 are connected with ~160 ohms Pin's 2,3 are connected with ~1400 ohms On my Plantronics head set. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: January 17, 2006 10:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Slightly OT: Plantronics headset quick connectorwiring Does anyone know where the order of the wires on this connector can be found? ___ --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] IBM eServers?
Have you considered the Sangoma cards? I have an a102 running in 2x X306's and they're running fantastic. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Harry McGregor Sent: December 19, 2005 4:20 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IBM eServers? Hi, Has anyone used a Digium PRI card in an IBM eServer x346? I know that Digium's website lists the x345 as having problems, but I am restricted to buying only IBM eServers for this possible project. I would like to use the TE411P Harry ___ --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] no sound with red alarm?
I have had no issues where asterisk is affected by a Sangoma card being down. I ran my test server like that for a few weeks doing lots of testing before I brought it up with a dummy card. Even now, if it's up or down it doesn't matter to asterisk. Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon EstepSent: August 25, 2005 10:20 PMTo: Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] no sound with red alarm? Does it make sense that a system with a single PRI (sangoma card) would loose its ability to play sounds when not connected to the PSTN for clock? Is there a way to configure ztdummy as a BACKUP clock source, or is there a better way altogether? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo calibration with ztmonitor and a testlinefrom a telco
The value of 14800 is correct. I had issues with my TDM400p with 2x FXO's installed and using the Xlite client. I could not get the echo stable at the initial call. Changing to a hard phone made everything work correctly. I still had problems with the off location I called, but mostly worked great. To see the TX gain I created an extension with the Milliwatt command attached to it and called it from the PSTN, then on a second line I called the milliwatt line at my CO and compared the volumes by ear (instead of looping around) and it worked great. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken Dresdell Sent: August 16, 2005 9:55 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Echo calibration with ztmonitor and a testlinefrom a telco Thanks for your help, I have already seen this page but since the head version of ztmonitor is able to show the real number value of the rx and tx (ztmonito -vv), I was thinking that maybe someone could confirm to which value we want the rx of ztmonitor when we try to calibrate the system with a test line from a Telco and a TDM card. The only information that I have found is that I have to setup my rxgain to get a 14800 value with ztmonitor but that is not working. Any others suggestions ? Regards Ken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Littlejohn Sent: 15 août 2005 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Echo calibration with ztmonitor and a test linefrom a telco On 8/15/05, Ken Dresdell [EMAIL PROTECTED] wrote: Hello everyone, Does anyone have experience with echo calibration for TDM card with rxgain and txgain (Zapata.conf) and ztmonitor (CVS Head)? I have found very few information about it and what I have found makes me confused. I have a phone number provided by my TelCo(1004 hz at 0db) and from what I saw, I am supposed to calibrate my rxgain to get a 14800 value with ztmonitor . Here is the information I found: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.ht ml Does anyone have successfully reduced echo with this procedure? My main problem is that when I get 14800 with ztmonitor, I have now a rxgain=14 and it seem to be too high for asterisk and I cannot dial out anymore. Any suggestions? Thanks in advance for your pointers Regards Ken ___ 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 I have been doing a bit of this too lately. This was also useful. http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/ docs-html/x1695.html Dan ___ 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] quad t1 / 1U rack server combos
From what I understand (From Sangoma's tech support) and having a IBM x306 SCSI system with an A102u I believe that the system will scale up to 4xT1's easily. With a full T1 of traffic coming in and playing music on hold, theCPU was at 7% with no transcoding. Sangomacards are supposed to place less draw on the interrupts and offer some new direct writing to DMA in their A104 cards. You may want to give them a call (Scott or Nenad are the two best people to speak with). From Sangoma README.asterisk: Voice data is channelized and grouped into 8 byte chunks in HARDWARE. Each voice channel is then DMAed directly into the ZAPTEL buffers. Thus there is ZERO copy from HARDWARE to ZAPTEL, resulting in better performance and scalability. It sounds to me like that would be once advantage over Digiums cards. They also have Hardware PRI functions that are passed directly to libpri. http://sangoma.com/linux/README.asterisk Hope that helps. Chad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon EstepSent: August 16, 2005 12:33 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] quad t1 / 1U rack server combos It is amazing to me at this point that there is not an official Digium list of supported servers (including 1u models!). Clearly the number 1 issue with the Digium PRI cards is the server that they are used in. The new cards even go as far as listing server that DO NOT work on the Digium site! The wiki references are old and do not have any testing parameters. Cmon guys! Certify a few current model servers and be done with it. Without that information I must again ask the question; What 1u server combos work with the new quad pri cards UNDER LOAD (more than 75% channel use). Every user that buys a Digium PRI card should not have to play hit or miss with 2 or 3 servers that cost more than the card to get it to work. Please Please Please publish something useful to support the sale of PRI cards. Damon ___ 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] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone
To use the old phones and existing wiring you'll need some E1/T1 FXS Channel banks and a T1/E1 Card. Each bank will handle 30/24 phones and pipe them into a single E1/T1 connection. You can connect up to 4 (or 8 soon from Sangoma) T1's per card. I really like the Sangoma cards, there are also Digium cards as well. The Wiki will have a lot more information regarding Channel Banks and FXS adapters, I would suggest starting there. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Rima Sent: August 11, 2005 8:34 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone I have a brief from a local hotel to build a PBX using Asterisk but they want to use their exisiting telephones and wiring from an old PBX that no longer works. Basically, I can build the system but an looking for a card that will allow for upto 20 extensions to be wired into the back of the PC. Doeas anyone know of a solution to this Sean-- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie ___ 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] Polycom 501 indicated -1 Urgent and 1 new for new voice mail
Title: Message Has anyone else seen this problem? MWI works, but when you press the messages button the display shows -1 urgent, 1 new, and 0 old. Anyone know how to fix this? Chad ___ 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] Looking for Thai DIDs
I do not know about Thailand DID's, but I would rather not see you post six times about this. There should be some information in the Wiki about providers all across the world and Google may have some additional information. Try the -biz list for biz' related questions. Or, the Wiki has a lot of information: http://www.voip-info.org/tiki-index.php?page=VOIP%20Service%20Providers% 20by%20Country -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Coulthurst Sent: July 21, 2005 1:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Looking for Thai DIDs Anybody know where to find Thailand DIDs that can ring in to my * in the USA on SIP? Oh, and a good price, too! ;) ___ 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] TDM04B - Takes long to initialize...
Remove Callerid and set immediate=yes Callerid is sent between the first and second rings, so asterisk has to wait for it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nitesh Divecha Sent: Monday, July 18, 2005 5:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TDM04B - Takes long to initialize... Hello All, I got my TDM04B card installed and configured. Everything works fine I can receive calls and route to appropriate extensions. The only problem I am facing is Slowness. When I dial the PSTN number which is connected to Zap 1-1 after two ring it answers and then run the AGI script. What I did was assign it to a specific extension. So all inbound call on that PSTN number should ring on that extension. But its too slow... First Zap 1-1 takes time to initialize and then transferring the call to extension takes time. Is there any way to speed up the process or reduce the timings, I can not find any info. Please help... Thanks Neel ___ 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 Auto-Answer problems
Ipmid still is being processed, sip.cfg contained the same information. I've removed it just to clean things up. Setting the class to the correct value solved the problem, I can't believe that I missed it. Thanks, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dbruce Sent: July 15, 2005 6:39 AM To: C F; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom Auto-Answer problems Although the ipmid.cfg has been deprecated with SIP v1.5.2 (all the parameters have been moved to sip.cfg), the firmware will still parse and use the ipmid.cfg file until you specifically update your existing configuration files. If you have already updated the configuration files, then both of the parameters will be in the sip.cfg file. Regards, Derek - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 14, 2005 9:04 PM Subject: Re: [Asterisk-Users] Polycom Auto-Answer problems The IP-501 AFAIK comes shipped with SIP1.5.2 which does NOT use ipmid.cfg. You have to get new *.cfg files for the ip-501 or the older phones that run 1.5.2. On 7/14/05, Chad Osmond [EMAIL PROTECTED] 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 alertInfo voIpProt.SIP.alertInfo.2.value=Ring_Ans voIpProt.SIP.alertInfo.1.class=4/ Ipmid.cfg RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=1000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ 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 ___ 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] Polycom configs?
But once you have one setup you can just buy a barcode scanner, scan the MAC from the label, print your secret and other data entries as barcodes and use the script to set them all up. I'm loving the polycom setup at this point, the central configuration setup is fantastic. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ted Serreyn Sent: July 15, 2005 2:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom configs? Ditto, I only have a couple of the polycomm phones spent the better part of 1 day figuring out how to get them configured properly. -- Ted Serreyn Phone:262-432-0260 Fax:262-432-0232 Serreyn Network Services, LLChttp://www.serreyn.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Friday, July 15, 2005 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom configs? Chris, as I look over my stack of unopened Polycom 501s, I think to myself that I would enjoy seeing your provisioning script if you wouldn't mind sharing it. Chris Mason (Lists) wrote: Michael Graves wrote: I have a number of Polycom phones to setup with my * server. For my initial few phones I hand wrote configs. Does anyone here who uses Polycom phones have some form of management utility for automating their setup? I wrote myself a very simple script that makes provisioning the phone a one line command. Let me know if you would like it. ___ 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] Polycom Auto-Answer problems
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 alertInfo voIpProt.SIP.alertInfo.2.value=Ring_Ans voIpProt.SIP.alertInfo.1.class=4/ Ipmid.cfg RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=1000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ 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 ___ 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] converting windows .wav to .gsm
Can you be a bit more specific as to what the problems is? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mohammad Sent: July 6, 2005 2:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] converting windows .wav to .gsm HI ALL; I have problem converting a windows .wav file to .gsm format by Sox. Could anyone help. Cheers, Mohammad ___ 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] zaptel missing /dev/zap after FC3 update
Did you try tailing the /var/log/dmesg to see what happened when you loaded zaptel and wctdm with modprobe? Check that /etc/modprobe.conf still contains the correct module entries. Does /lib/modules/2.6.11-1.35_FC3smp/misc still contain and correct wctdm.ko files? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Howard Ratzlaff Sent: July 6, 2005 1:57 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] zaptel missing /dev/zap after FC3 update I have a P4 HT system running * on a 2.6.9-1.667smp kernel (Fedora Core 3). Everythng was testing out and the configuration was working. After running YUM update, kernel 2.6.11-1.35_FC3smp was installed. Now Zaptel cannot find /dev/zap. Waiting for zap to come online...Error: missing /dev/zap! I have already recompiled zaptel, libpri, and asterisk after changing the /usr/src/linux-2.6 symbolic link (linux-2.6 - /lib/modules/2.6.11-1.35_FC3smp/build/). There is only a TDM22b installed I reverted to the older kernel, recompiled and have the same issue. Any thoughts? ___ 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] Re: teliax [Was: LiveVoip is Bankrupt]
For someone that places outbound calls only, in a fairly low volume, is there a recommendation for which one would be best for me? I have had continual audio trouble with LiveVOIP, though other services (FWD) work fine, so I'd want something that has good audio quality. I will toss in my $0.02 and say that I have had good luck with Voxee, simple setup, good quality, not so great instructions (there weren't any) but a snappy response from the time I paid in paypal until the time it appeared on my account (2 minutes) Their rates seem good as well, I'm happy with them. There was an issue with their servers ignoring the CID information, but that has been resolved. ___ 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] Zaptel HEAD with * Stable?
Yes, I'm running it right now, CVS as of a few days ago, and * 1.0.7 on 2.6.x kernel and FC2. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: June 20, 2005 2:44 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Zaptel HEAD with * Stable? Will the CVS HEAD version of the Zaptel drivers work with the STABLE branch of *? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ 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] choice of processors
The Dual 2.8GHz will be much faster for running everything. If it is the same price it should be a no brainier, take the two CPU system. Depending on the manufacture of the system it may even take a failure of one CPU. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Langley Sent: May 30, 2005 11:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] choice of processors Hi there I am moving into a production environment. I will mostly be using Meetme, with Ztdummy for timing. I have a question on which of 2 processor setups is favourable. I have the choice between Dual 2.8GHz Xeon Processors and a single Pentium 4 3.06GHz Processor. These will cost me exactly the same amount. Would one of these processor setups be favourable, both in terms of performance and running Asterisk? Many thanks Steven ___ 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] OT: DSL problems - UNREACHABLE - REACHABLE
Welcome to DSL, the telco didn't do any more tests then required to get sync for 30 seconds. Cancel the DSL and get another line. That's about the extent of it, or at least in Ontario it is, I've had this problem with 5 or 6 connections. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: May 2, 2005 10:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] OT: DSL problems - UNREACHABLE - REACHABLE I could really use some input here, forgive the OT nature, but my problem is related to asterisk and voIP on a DSL connection and becoming a big mystery. I noticed about three weeks ago a lot of UNREACHABLEs that became REACHABLE 10 seconds later. After studying this a little, it happens that the DSL connection was stopping every 8 minutes (+ about 3 seconds). The modem doesn't apperat to lose sync, the data flow just stops. Since then I've removed asterisk from that connection. Every possible test has been done at our office, three different modem/routers of different brands were swapped in/out, there is a second phone line in the same cable that is on a different connection and it does not have the interruptions. I've turned off every box in the office and disconnected every cable from the router.Also disconnected FXO lines, phones and left just a modem/router on. No change. The 8 minutes are invariable, so after turning of everything here, I can't see how it could possibly be any local hardware. The phone company here has, after being evasive aboput checking the DSLAM, claimed they did everything possible, changed our DSLAM connection, tried every piece of equipment on their end. Ditto the ISP who has been very cooperative. I can only think of one more possible approach: get the power lines and the phone line independently checked for some kind of parasitic interference, say a big machine of some kind going on and off. Why this affects one DSL connection and not the other... I wouldn't know. Does anyone have any suggestions about what kind of outfit to look for that might do this kind of checking? Or any suggestions to pursue at all? tia ___ 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] Incoming Not Answering
-Original Message- snipped From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Sampson -- Executing Answer(Zap/1-1, ) in new stack -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Apr 25 17:01:15 NOTICE[3515]: chan_zap.c:5374 ss_thread: Got event 2 (Ring/Answered)... -- Executing Wait(Zap/1-1, 1) in new stack -- Executing Answer(Zap/1-1, ) in new stack -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' === Hi David, Can you post your extensions.conf file, there may be a clue somewhere in the exten = s, section. If you included the default example it should be working, but there may be something that has changed. Chad ___ 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
