[asterisk-users] TE220 PCI express performnace
Dear all I have going to put more PRI line in my organization theseday i have decide to put all PRI on TE220 dual span e1/t1 pci express card so what about the performnace and installation of this card is there anybody useing this card suggest me .?? PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video Call
Hi, Thereis any application (SIP) + Video can installed at phone, so with this application can commnication with asterisk to do video call Thanks On 11/4/07, Yann JOUANIN [EMAIL PROTECTED] wrote: Hi, A few time ago I read an article which explain how to use a 3G video phone with Asterisk. The article was in French bit the idea was : _using a BRI card (with modularISDN) and using h324m lib. yann -Message d'origine- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Bert Haverkamp Envoyé: samedi 3 novembre 2007 21:03 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: Re: [asterisk-users] Video Call 2007/11/1, voip Server asterisk [EMAIL PROTECTED]: Hi.. Iam new with asterisk PBX, and i have read about asterisk video call.: my question: 1. Is imposible to establish system video call (from Phone with GPRS/3G enabled to Computer Running Softphone like X-Lite) over Asterisk Gateway.. 2. If posible what requirement (Hardware and Software on my Asterisk,PC or My Phone) Thanks Joko Pitoyo This is generally not possible. The 3G phones (GPRS will be a strech wrt bandwidth) that do video telephony, do not support any SIP. So the operator will have to introduce a sort of SIP-3GPP interface box in his network. I currently do not know of any operator supporting this. Regards, Bert ___ --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 -- --- Bert en Selena www.bertenselena.net - There are 10 kind op people in the world: those who understand binary, and those who don't. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Not Hearing hello-world Play
Hi Asterisk Gurus! My lab asterisk box has 1 FXO and 1 FXS ports in it. I connect a GSM phone to the FXO port. I connect a regular corded phone to the FXS port. The Dial() application for both incoming and outgoing calls specifies the A(hello-world) flag. From another GSM phone, if I call the extension (corded) phone attached to the box, it plays the hello-world file when I pick it up. But from the extension phone if I dial to another GSM phone and I pick it up, it does not play (or at least I don't hear it play) the hello-world. What's the obvious thing I'm overlooking here? Thanks, Jeng ___ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE220 PCI express performnace
On Mon, Nov 05, 2007 at 12:10:45AM -0800, satish patel wrote: Dear all I have going to put more PRI line in my organization theseday. i have decide to put all PRI on TE220 dual span e1/t1 pci express card so what about the performnace and installation of this card is there anybody useing this card suggest me .?? Just some obvious follow-ups: * Do you you currently have some PRI lines? * How many do you have? How many will you have? * Do you currently use some card / adapter? Using some punctuation in your message (e.g: a dot in the end of a sentence) can make your message clearer. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need Reference sites
Hi, Various site available for asterisk,listed below, www.asterisk.org www.voip-info.com www.digium.com and best is search in www.google.com On Nov 5, 2007 5:22 AM, Michael Davidson [EMAIL PROTECTED] wrote: Hi, I'am comparative newbie to the world of Asterisk. I'd like to introduce an Asterisk based PBX into my company but need to convince my executive of it's worthiness. I need some reference sites to quote in my discussion, preferably well known companies of course. I have surfed the net but not come up with anything of note, if anyone can help it would be greatly appreciated. Thanks, Mike D. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Read too short
On 11/3/07, John Faubion [EMAIL PROTECTED] wrote: Am I the *ONLY* one that has this issue? John Faubion -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Faubion Sent: Thursday, November 01, 2007 11:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTP Read too short Hello, I'm getting the following logs: [Nov 1 10:54:37] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short [Nov 1 10:54:39] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short [Nov 1 10:54:40] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short Anyone know how to correct this? I'm using SIPConnect from CBeyond and this appears on incoming calls. I haven't had any complaints about voice quality and I haven't seen any dropped calls. Should I be concerned? There are several cases of this discussed on the list, so you can search the archives. The basic answer is that it is telling the truth. The usual resolution seems to be that a copy of Wireshark, and a basic understanding of RTP is necessary to diagnose the problem. I'm afraid I cannot help you there :) Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 40, Issue 5
i dont know how to remove these errors. But i think you should try asterisk-addons package available from asterisk download site. it contains the h323 channel also. You only need to compile it. remove the asterisk-oh323 package from your system and install the asterisk-addons package. I hope this solves your problem. On Nov 5, 2007 8:42 AM, Bincy K. Philip [EMAIL PROTECTED] wrote: Hello Thanks for the reply.. I could use Asterisk as SIP server and establish call using two SIP phones. But I need H323 support also. For that I have compiled the files in asterisk/channel/h323 and installed without problem. But even after i have started Asterisk,it is not supporting h323 commands like h323 debug,h323 show codecs. So i tried to install compile asterisk-oh323. i got an error that channel_pvt.h is missing..when i downloaded and put the same file i got double declaration error. I have excluded channel_pvt.h from chan_oh323.c include file list, but got errors. Anyone please help! Thanks Regards Bincy K Philip Date: Fri, 2 Nov 2007 17:50:57 +0500 From: Rizwan Hisham [EMAIL PROTECTED] Subject: Re: [asterisk-users] asterisk as a gateway To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Hi, You should visit the following websites for help www.voip-info.org www.asteriskguru.com www.nerdvittles.com But the best step for beginners is to read the Asterisk, The Future of Telephony book which is available freely on asterisk website. It will help you great deal in understanding basics of asterisk. Im not sure about h323 but the book will help you to add some contents in extensions.conf. You can start with sip.conf instead coz its help is provided in the book. On Nov 2, 2007 2:26 PM, Bincy K. Philip [EMAIL PROTECTED] wrote: Hello, Could anyone please give some information on configuring asterisk as a gateway. What contents have to add in h.323 .conf and extensions.conf files ? Thanks Regards Bincy K Philip ___ --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 -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to disable Asterisk 407 Proxy Authentication Required Challenge response
Hi, I have an UAC registered in VoIP provider. (register command in sip.conf) When I try to make call from PSTN through this VoIP provider, when INVITE reaches asterisk is sents 407 Proxy Authentication Required Challenge response. How can I disable this, because I want to allow any external call from my sip provider. Thanks in advance Tomasz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk as a gateway
Thanks once again..I will check with addon package and let you know the status.. Date: Mon, 5 Nov 2007 15:30:49 +0500 From: Rizwan Hisham [EMAIL PROTECTED] Subject: Re: [asterisk-users] asterisk-users Digest, Vol 40, Issue 5 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 i dont know how to remove these errors. But i think you should try asterisk-addons package available from asterisk download site. it contains the h323 channel also. You only need to compile it. remove the asterisk-oh323 package from your system and install the asterisk-addons package. I hope this solves your problem. On Nov 5, 2007 8:42 AM, Bincy K. Philip [EMAIL PROTECTED] wrote: Hello Thanks for the reply.. I could use Asterisk as SIP server and establish call using two SIP phones. But I need H323 support also. For that I have compiled the files in asterisk/channel/h323 and installed without problem. But even after i have started Asterisk,it is not supporting h323 commands like h323 debug,h323 show codecs. So i tried to install compile asterisk-oh323. i got an error that channel_pvt.h is missing..when i downloaded and put the same file i got double declaration error. I have excluded channel_pvt.h from chan_oh323.c include file list, but got errors. Anyone please help! Thanks Regards Bincy K Philip -- Message: 8 Date: Mon, 05 Nov 2007 01:52:24 +0200 From: Michael Davidson [EMAIL PROTECTED] Subject: [asterisk-users] Need Reference sites To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, I'am comparative newbie to the world of Asterisk. I'd like to introduce an Asterisk based PBX into my company but need to convince my executive of it's worthiness. I need some reference sites to quote in my discussion, preferably well known companies of course. I have surfed the net but not come up with anything of note, if anyone can help it would be greatly appreciated. Thanks, Mike D. -- Message: 9 Date: Mon, 05 Nov 2007 11:17:39 +1100 From: Paul Hales [EMAIL PROTECTED] Subject: Re: [asterisk-users] 7960 Queue Issue To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain My memory tells me that there is a flag (something like 'ringinuse') which can make sure this sort of thing does not happen. PaulH On Mon, 2007-11-05 at 10:26 +1100, Nick Brown wrote: Morning All, Quick question that has me stumped. Have a queue with several members (Statically defined in queues.conf at this stage for testing) who use Cisco 7960's. The queue is configured to use rrmemory and generally this works correctly. However if a member is already on a call their phone will still ring (The 7960 can show multiple incoming calls for one line). I really don't want members who are on calls to get more calls. Especially when we start logging out members who don't answer. Asterisk shows; -- Called 1014 -- SIP/1014-08f2e4d0 is ringing -- Local/[EMAIL PROTECTED];1 is ringing -- Nobody picked up in 15000 ms Short of disabling the feature to show multiple incoming calls on the 7960's (Which I don't know if it can be done anyway), has anyone got any suggestions? Thanks in advance! Nick. ___ --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 -- Message: 10 Date: Mon, 5 Nov 2007 00:51:10 + From: Frank Church [EMAIL PROTECTED] Subject: [asterisk-users] Are the ATAs which can allow multiple extensions from one network connection? To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Are there ATAs that allow different phone numbers from one network connection? Such as supporting multiple IP addresses so that each RJ11 has a different extension or some other way? -- Message: 11 Date: Sun, 4 Nov 2007 19:57:07 -0500 From: Eric Merkel [EMAIL PROTECTED] Subject: Re: [asterisk-users] 7960 Queue Issue To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 On 11/4/07, Nick Brown [EMAIL PROTECTED] wrote: Morning All, Quick question that has me stumped. Have a queue with several members (Statically defined in queues.conf at this stage for testing) who use Cisco 7960's. The queue is configured to use rrmemory and generally this works correctly.
[asterisk-users] How to delete voice mail messages?
Hello all, Could I create a script to delete the first messages on my voice mail? In this script should I update any messages index file or there isn't any file to index them? Could you share any script to do that? Thanks in advance. VoipCrazy. ___ --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] How to delete voice mail messages?
On 12:15, Mon 05 Nov 07, voip crazy wrote: Hello all, Could I create a script to delete the first messages on my voice mail? In this script should I update any messages index file or there isn't any file to index them? Could you share any script to do that? Hi, Voicemails are stored in /var/spool/asterisk/voicemail/context/vmbox by default. There's some .wav files and a .txt file for every message. You can easily delete them using some shellscript. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which Variable???
Hi Gurus! Please excuse this pesky Asterisk rookie:-) I just wanted to know which channel variable tells asterisk the number of rings before an incoming call on FXO channel is answered? I looked through zapata.conf.sample and other places and could not find something there readily. Thanks, Jeng ___ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good http://uk.promotions.yahoo.com/forgood/environment.html ___ --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] Dynamic Queue Members - Auto Logoff
You can use RemoveQueueMember(queuename) to dynamically remove the agents. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Brown Sent: Sunday, November 04, 2007 11:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dynamic Queue Members - Auto Logoff Another quick question (Spending the day trying to get this project sorted and tucked away) If I am dynamically adding queue members, they will not abide to settings within agents.conf will they? Ie. I need the equivalent of Autologoff however want my agents to receive calls when someone joins the queue, not have to sit on hold all day. I see AgentCallbackLogin has finally been removed. Has anyone got a work around for this? Thanks. Nick. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need Reference sites
We have implemented asterisk. We are a tier one automotive tooling supplier. We have approx. 120 extensions in use plus 8 fax extensions. We also have a two port cell phone adapter so when we call out T-Mobile phone, we are using the free (included) T-mobile to T-mobile minutes. We have also added an IAX2 activeX control to our website, so that people can call us for free. This was originally added for overseas JVs to call us without cost. Notes for cost savings: We kept the main panel (was a 3 panel system) of our old PBX to use for its analog extensions for faxing. This lets us keep our fax lines as TDM, so there are no fax over IP issues. We use a PRI cable to connect the asterisk and old PBX. We used Citel digital extension converters to reuse our old NEC phones as SIP extensions. This made our phone costs one third of what it would cost to buy new IP phones. The negative here is that we still have to maintain our two wire extension cables. We did this change mostly for the feature set. We do have a few IP phones (wireless and wired) for areas of our shop that did not have two wire phone lines. We currently have consolidated down to one building, but when we next expand, a VOIP phone system will really shine in its cost savings. I did this all myself. It was not too difficult once you get into it. We started off by dropping the asterisk server between the Telco and old PBX using PRI. We then converted over to asterisk's voice mail, CDR and conferencing (web-meetme). We ran like this for a few months with only my extension on the new system. (for testing and adding features) We had a catch-all rule in asterisk where all _5XXX calls were forwarded to the old PBX. When we started the rollout: We just defined the extension in asterisk so inbound calls would stay in asterisk and go to the new extension. We removed the extension number from the old PBX, so calls to the extension from the old PBX would be dialed as an outside call and go to asterisk. This allowed us to take our time rolling out the new system. We moved maybe 1 or 2 extensions a day and could do it during work hours. Our total purchased cost was $9000. I did not track my time exclusive to this project, so I do not have a dollar figure for it. Much of this time was fun for me though. I actually had to change the source code (I am not a programmer and do not know how to write C, but I can read and alter code OK). Our old PBX would hang up if we tried to send callerID. Our telco at the time was sending only the number, but asterisk was send number and name. I added a config option to allow for this and it was added to the source code at Digium. Now we have changed Telcos and we have name and number in asterisk and only number at the fax machines, which is fine. At this point, we are running asterisk with no source code changes. (this makes upgrades easier) I hope this helps. -- -- Steven http://www.glimasoutheast.org Michael Davidson [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, I'am comparative newbie to the world of Asterisk. I'd like to introduce an Asterisk based PBX into my company but need to convince my executive of it's worthiness. I need some reference sites to quote in my discussion, preferably well known companies of course. I have surfed the net but not come up with anything of note, if anyone can help it would be greatly appreciated. Thanks, Mike D. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compatibility Issues with dell poweredge 195and TE110P card
2950s work fine. I have had the parity error for over a year with no noticable problems. It is working fine. I did have to make some IRQ changes to clean up the system. I did these on my Dell 1750 test machine, but have made the same changes on my production machine. The changes basically redue the IRQ load from other cards, like the RAID card, which will reduce the bus's capacity for processing all of the TDM IRQs. It also allocates just one CPU full time for all of the TDM IRQs. The changes are below: ref: FYI on zttool output on SMP system --- Results after 56 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.999564 Only 2 were 99.987793, the 54 others were all 100.00. I got this by making the changes below on my dual proc Dell 1750. setpci -v -s 01:08.1 LATENCY_TIMER=8 setpci -v -s 00:0f.1 LATENCY_TIMER=8 setpci -v -s 01:04.0 LATENCY_TIMER=8 setpci -v -s 01:02.0 LATENCY_TIMER=8 setpci -v -s 00:0f.2 LATENCY_TIMER=8 setpci -v -s 01:04.0 LATENCY_TIMER=8 (these are USB, SCSI HW RAID driver, Ethernet, Video, etc. I did not alter ZAP cards, nor any bridges or buses) echo 1 /proc/irq/17/smp_affinity (Ethernet) echo 1 /proc/irq/18/smp_affinity (SCSI HW RAID Driver) echo 2 /proc/irq/20/smp_affinity (TDM) echo 2 /proc/irq/24/smp_affinity (TE411P) I also turned of the startup of irqbalance. The setpci changes did the most work concerning reaching 100% in zttest. Irqbalance was causing the the processor handling the interrupts of the zap cards to change very often. This would impose a delay during the change and cause the zttest numbers to drop/be inconsistent. Because I turned irqbalance off, the irqs are processed round robin style, which is also not good. Therefore, I hard coded the processor affinity for the zap cards to one proc and all other high load irqs to the other proc. If you have more than 2 procs, you can spread them out even more. If you do not turn off irqbalance, the affinity changes will be overwritten by it. I made these changes on a live system without issue. I set these changes in /etc/rc.d/rc.local to reset them after reboots. -- -- Steven http://www.glimasoutheast.org Brian Hutchinson [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] You guys are scaring me! I'm building a 2950 with SAS RAID 5 on the new PERC and it will have 2 TE420P's. I hope it works or my bacon will fry. On 10/25/07, Joseph Begumisa [EMAIL PROTECTED] wrote: Has anyone had any compatibility issues with a TE110P card installed on a Dell Poweredge 1950?I noted the following error on the LCD display of the Dell Poweredge 1950: E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0. Yes, I have had this problem with a dell PE1650, 1850, SC1400, and PE650. I have a TE410P that does it. It may not be wise, but I just ignore the orange blinking LCD display (or light, depending on the model). I did try reseating the card, and it works for a few weeks, and then goes back to the same old thing. Yes, that happened too. Digium has graciously offered to send me a TE120P with the Digium VoiceBus technology which I will test out on the Dell 1950. Will post my findings thereafter. Joseph. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which Variable???
Jeng Yu wrote: Hi Gurus! Please excuse this pesky Asterisk rookie:-) I just wanted to know which channel variable tells asterisk the number of rings before an incoming call on FXO channel is answered? I looked through zapata.conf.sample and other places and could not find something there readily. There is no such setting. Asterisk will answer as soon as the call comes in. If Asterisk is configured for Caller*ID, then Asterisk will wait until the 2nd ring (waiting for Caller*ID information to arrive). ___ --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] Which Variable???
Jeng Yu wrote: Hi Gurus! Please excuse this pesky Asterisk rookie:-) I just wanted to know which channel variable tells asterisk the number of rings before an incoming call on FXO channel is answered? I looked through zapata.conf.sample and other places and could not find something there readily. Thanks, Jeng To my knowledge, there isn't one. There are two things that affect how an FXO channel is answered. The first one is whether usecallerid is set on the channel. If it is set to yes on a channel that doesn't actually have caller ID it can cause a roughly 2 ring delay before the call is passed to the dial plan. The second, is based entirely on the dial plan. When a call comes in and is passed to an extension in a certain context it's up to you when/how/if the call is ever answered. -Dave ___ --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] Dynamic Queue Members - Auto Logoff
On 11/5/07, Nick Brown [EMAIL PROTECTED] wrote:
Another quick question (Spending the day trying to get this project sorted
and tucked away) If I am dynamically adding queue members, they will not
abide to settings within agents.conf will they?
correct.
Ie. I need the equivalent of Autologoff however want my agents to receive
calls when someone joins the queue, not have to sit on hold all day. I see
AgentCallbackLogin has finally been removed.
Has anyone got a work around for this?
It hasn't been removed (in 1.4), just deprecated (I assume you're not trying
this with -trunk). Still, it's not compatible with adding members via
AddQueueMember().
There is an example of doing auto-logoff in docs/queues-
with-callbackmembers.txt in the source distribution. Look for macro
callagent for the specific block that does the work. You do have to be
using Local channels to make this work though, as you need to Dial() the
actual device from the dialplan, then check ${DIALSTATUS} to make decisions
about what to do if the agent doesn't pick up.
j.
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Re: [asterisk-users] Kirk IP600/3 Wireless Server SIP config
On Fri, 26 Oct 2007, Benny Amorsen wrote: RB == Remco Barendse [EMAIL PROTECTED] writes: RB Hi list! Is anyone using the Kirk IP600/3 with SIP firmware RB connected to Asterisk? Yes. RB If anyone would be willing to share the dump of their IP600 config RB file, i would really appreciate it. Sorry I'm not at work right now. If I get time later, I will. Hi Benny! Did you manage to make a dump of a working configuration from the IP600/3? Would be really useful, can't seem to get it to work properly. Thanks! Remco ___ --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] OT: Which SIP method to use for this specific behaviour ?
Hello, Let SIP extensions 1001 and 1002 belong to an Asterisk calling group : whenever an coming call reaches this calling group, both extensions 1001 and 1002 receive a SIP INVITE message which makes these 2 phones starting to ring. When a callee picks up his phone, the other extension receives a CANCEL or BYE message which stops ringing. Is there any option you can include in CANCEL or BYE messages so that the SIP hardphones would understand it shouldn't have to log this call as it has been replied by someone else ? In other words, is there any Alert-info option which can be used to pilot phones call history logs ? I didn't dare to search myself in IETF archives, given the number of standards, SIP is now including. Regards ___ --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] Parameters effect on the success registeration
Hi All; nat=yes for example, it effects on the success of the registeration. What are the parameters that might let the registeration fail when I need to register Asterisk on a softswitch using register = ? Any help? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Which SIP method to use for this specificbehaviour ?
Search for: Reason: SIP ;cause=200 ;text=Call completed elsewhere From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: 05 November 2007 15:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT: Which SIP method to use for this specificbehaviour ? Hello, Let SIP extensions 1001 and 1002 belong to an Asterisk calling group : whenever an coming call reaches this calling group, both extensions 1001 and 1002 receive a SIP INVITE message which makes these 2 phones starting to ring. When a callee picks up his phone, the other extension receives a CANCEL or BYE message which stops ringing. Is there any option you can include in CANCEL or BYE messages so that the SIP hardphones would understand it shouldn't have to log this call as it has been replied by someone else ? In other words, is there any Alert-info option which can be used to pilot phones call history logs ? I didn't dare to search myself in IETF archives, given the number of standards, SIP is now including. Regards ___ --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] OT: Which SIP method to use for this specificbehaviour ?
Thanks for the tip. If I may ask, do you if this signaling is support in Asterisk 1.4 ? 2007/11/5, Steve Langstaff [EMAIL PROTECTED]: Search for: Reason: SIP ;cause=200 ;text=Call completed elsewhere -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Olivier *Sent:* 05 November 2007 15:22 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] OT: Which SIP method to use for this specificbehaviour ? Hello, Let SIP extensions 1001 and 1002 belong to an Asterisk calling group : whenever an coming call reaches this calling group, both extensions 1001 and 1002 receive a SIP INVITE message which makes these 2 phones starting to ring. When a callee picks up his phone, the other extension receives a CANCEL or BYE message which stops ringing. Is there any option you can include in CANCEL or BYE messages so that the SIP hardphones would understand it shouldn't have to log this call as it has been replied by someone else ? In other words, is there any Alert-info option which can be used to pilot phones call history logs ? I didn't dare to search myself in IETF archives, given the number of standards, SIP is now including. Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk versions and H323
Dear Dovid; Thanks a lot for the nice reply and support. I need a document on this addon (file name to be downloaded, steps to compile, where i can find the h323 module in this addon, and the configuration for h323)? Regards Bilal There is a version in the asterisk add-ons that is fairly simple to use. - Original Message - From: bilal ghayyad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, November 03, 2007 10:43 PM Subject: [asterisk-users] Asterisk versions and H323 Hi List; Is there an Asterisk version that contains H323 module, or still I have to download the h323 alone and compile it? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which Variable???
On Mon, 5 Nov 2007, Eric ManxPower Wieling wrote: Jeng Yu wrote: Hi Gurus! Please excuse this pesky Asterisk rookie:-) I just wanted to know which channel variable tells asterisk the number of rings before an incoming call on FXO channel is answered? I looked through zapata.conf.sample and other places and could not find something there readily. There is no such setting. Asterisk will answer as soon as the call comes in. If Asterisk is configured for Caller*ID, then Asterisk will wait until the 2nd ring (waiting for Caller*ID information to arrive). Not neccessarily true in the UK where the caller ID is transmitted before the first ring. If you want to delay answering, then you can use the Wait instruction in the dialplan, but it's caligrated in seconds not rings. (Which are highly country dependant anyway) Gordon ___ --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] Meetme - how to protect the conference?
Hi all, I am just wondering - it there any way how to protect a conference from being abused by someone? I know I can request pin, but that pin is then hardcoded in meetme.conf and normal user can not change it. I would like to establish an admin user who could set a pin for the conference to be used by other participants. Is that possible? Thanks, Ondrej The information contained in this e-mail and in any attachments is confidential and is designated solely for the attention of the intended recipient(s). If you are not an intended recipient, you must not use, disclose, copy, distribute or retain this e-mail or any part thereof. If you have received this e-mail in error, please notify the sender by return e-mail and delete all copies of this e-mail from your computer system(s). Please direct any additional queries to: [EMAIL PROTECTED] Thank You. Silicon and Software Systems Limited. Registered in Ireland no. 378073. Registered Office: Whelan House, South County Business Park, Leopardstown, Dublin 18 ___ --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] Problem with CDR userfield not being set
I'm trying to use the MySQL CDR records.
According to dialplan show, the line in the dialplan is:
11. Set(CDR(userfield)=${billing_code}) [pbx_ael]
It looks like the value is being set when I watch the console during the call:
-- Executing [EMAIL PROTECTED]:11] Set(SIP/icall-0075a2e0,
CDR(userfield)=boatmenu) in new stack
But the record that's created in mysql has a blank userfield:
INSERT INTO `cdr` (`calldate`, `clid`, `src`, `dst`, `dcontext`,
`channel`, `dstchannel`, `lastapp`, `lastdata`, `duration`, `billsec`,
`disposition`, `amaflags`, `accountcode`, `uniqueid`, `userfield`)
VALUES \
('2007-11-05
17:25:17','(removed)','(removed)','s','restphone_event_loop','SIP/icall-0075a2e0','','Read','Result|/var/lib/asterisk/sounds/restphone_cepstral/016d4fda5256dc9a944d7102fac4',25,15,'ANSWERED',3,'1\
','','');
What am I missing? I'm running 1.4.13.
- James Moore
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Re: [asterisk-users] Meetme - how to protect the conference?
You could use meetme realtime and have the admin update the pin via a web interface instead. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ondrej Valousek Sent: Monday, November 05, 2007 09:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Meetme - how to protect the conference? Hi all, I am just wondering - it there any way how to protect a conference from being abused by someone? I know I can request pin, but that pin is then hardcoded in meetme.conf and normal user can not change it. I would like to establish an admin user who could set a pin for the conference to be used by other participants. Is that possible? Thanks, Ondrej ___ --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] Problem with CDR userfield not being set
On Mon, 2007-11-05 at 09:40 -0800, James Moore wrote:
I'm trying to use the MySQL CDR records.
According to dialplan show, the line in the dialplan is:
11. Set(CDR(userfield)=${billing_code}) [pbx_ael]
It looks like the value is being set when I watch the console during the call:
-- Executing [EMAIL PROTECTED]:11] Set(SIP/icall-0075a2e0,
CDR(userfield)=boatmenu) in new stack
But the record that's created in mysql has a blank userfield:
INSERT INTO `cdr` (`calldate`, `clid`, `src`, `dst`, `dcontext`,
`channel`, `dstchannel`, `lastapp`, `lastdata`, `duration`, `billsec`,
`disposition`, `amaflags`, `accountcode`, `uniqueid`, `userfield`)
VALUES \
('2007-11-05
17:25:17','(removed)','(removed)','s','restphone_event_loop','SIP/icall-0075a2e0','','Read','Result|/var/lib/asterisk/sounds/restphone_cepstral/016d4fda5256dc9a944d7102fac4',25,15,'ANSWERED',3,'1\
','','');
What am I missing? I'm running 1.4.13.
Do you have userfield=1 in your cdr_mysql.conf file?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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Re: [asterisk-users] Which Variable???
The call is not answered until you answer it with either the Answer app, or issuing a playback command etc. On 11/5/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Mon, 5 Nov 2007, Eric ManxPower Wieling wrote: Jeng Yu wrote: Hi Gurus! Please excuse this pesky Asterisk rookie:-) I just wanted to know which channel variable tells asterisk the number of rings before an incoming call on FXO channel is answered? I looked through zapata.conf.sample and other places and could not find something there readily. There is no such setting. Asterisk will answer as soon as the call comes in. If Asterisk is configured for Caller*ID, then Asterisk will wait until the 2nd ring (waiting for Caller*ID information to arrive). Not neccessarily true in the UK where the caller ID is transmitted before the first ring. If you want to delay answering, then you can use the Wait instruction in the dialplan, but it's caligrated in seconds not rings. (Which are highly country dependant anyway) Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Free T1 Card?
Gang, I recall several months ago that there was a company that was giving away a free 1-port T1 card, with some specific conditions. Do any of you recall who that was? My Google searches are coming up empty and now I'm wondering if I was hallucinating... Thanks, MC ___ --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] Testcall
# ./testcall testcall.conf Chan 1, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025860' to '013331339767' Chan 2, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025861' to '013331339768' Chan 3, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025862' to '013331339769' Chan 4, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025863' to '013331339770' Chan 5, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025864' to '013331339771' Chan 6, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025865' to '013331339772' Chan 7, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025866' to '013331339773' Chan 8, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025867' to '013331339774' Chan 9, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025868' to '013331339775' Chan 10, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025869' to '013331339776' Loading protocol mfcr2 Failed to open channel: Device or resource busy Why??? My testcall.conf is: destination-no 013331339767 protocol-class mfcr2 protocol-variant ar,10,4 protocol-end cpe caller no originating-no 30025860 on-offered accept circuits 1-10 Thanks!!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help: Static and dropped calls
Does anybody know why am getting a lot of static and sometimes dropped calls from my asterisk server. Vitelity is my number provider if it matters. Thank you Jarga Jallow image001.jpg___ --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] Arbitrary limit on length of email address?
I'm trying to get emailing of voicemail messages to work and by and large it does... However one email address is quite long in comparison to others I am testing and it fails to get delivered. For example - this one works and gets delivered: [Nov 5 18:35:14] DEBUG[2509]: app_voicemail.c:1957 sendmail: Sent mail to [EMAIL PROTECTED] with command '/usr/sbin/sendmail -t' And is 23 characters in length Whereas this one: [Nov 5 18:36:02] DEBUG[2519]: app_voicemail.c:1957 sendmail: Sent mail to [EMAIL PROTECTED] with command '/usr/sbin/sendmail -t' fails to get delivered and is 34 characters long. Both email accounts work otherwise and I have had no recorded problems with mails not arriving at the 34ch address before. Any ideas? Am I barking up the wrong tree? Cheers Alan ___ --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] AEX800 (TDM800 Express) - not detected
Mark J Elkins wrote: I have a AEX800 PCI Express card - sort of a TDM800 with PCI-Express. (or AEX844 - 4FXS 4FXO) It's not 'sort of', it *is* a TDM800 with a PCI Express bus interface. With only this card in the box Asterisknow gives me... no functional digium card found in /proc/zaptel - or words to that effect. This is a new card, and is probably not included in the version of Zaptel included in the most recently distributed AsteriskNOW ISO available from asterisknow.org. However, if you go to the rPath website and follow the links to the AsteriskNOW project, you can download an ISO image of 'beta 6.5', which includes updated components and likely will recognize your AEX800 card. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --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] Free T1 Card?
On Nov 5, 2007 1:14 PM, Michael Collins [EMAIL PROTECTED] wrote: Gang, I recall several months ago that there was a company that was giving away a free 1-port T1 card, with some specific conditions. Do any of you recall who that was? My Google searches are coming up empty and now I'm wondering if I was hallucinating… Thanks, MC http://www.pikatechnologies.com/ -- Kristian Kielhofner ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Arbitrary limit on length of email address?
Alan Lord wrote: Whereas this one: [Nov 5 18:36:02] DEBUG[2519]: app_voicemail.c:1957 sendmail: Sent [mail to [EMAIL PROTECTED] with command '/usr/sbin/sendmail -t' fails to get delivered and is 34 characters long. Both email accounts work otherwise and I have had no recorded problems with mails not arriving at the 34ch address before. Any ideas? Am I barking up the wrong tree? Check your mail-logs. Was the email with the long address accepted and processed by your mail-server? Also look for traces of an incomplete email-address being used (or something like that). /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --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] Arbitrary limit on length of email address?
Per Jessen wrote: snip / Check your mail-logs. Was the email with the long address accepted and processed by your mail-server? Also look for traces of an incomplete email-address being used (or something like that). /Per Jessen, Zürich Thanks Per, I checked my exim logs and that email address is being rejected by spamhaus because it's coming from an unauthenticated server, and it's on a dynamic IP address! Many thanks for making me look :-) Alan ___ --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] Problem with CDR userfield not being set
On 11/5/07, Carlos Chavez [EMAIL PROTECTED] wrote: Do you have userfield=1 in your cdr_mysql.conf file? Thanks - that took care of it. - James ___ --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] Meetme - how to protect the conference?
I am just wondering - it there any way how to protect a conference from being abused by someone? I know I can request pin, but that pin is then hardcoded in meetme.conf and normal user can not change it. I would like to establish an admin user who could set a pin for the conference to be used by other participants. Is that possible? Thanks, You can create dynamic conferences - with or without specifying conference PIN. The first one who joins conference enter's room number and PIN, that is then used for authentication. Of course you can prompt before for some extra code - to allow creation of conferences. Admins have other meaning for conferences - they can kick users, etc.. Regards, Atis ___ --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] RTP Read too short
I saw this with Grandstream GXP2000. When the phone is on a call and on mute, the phone sends SIP keepalive packets that are, indeed, too short. So asterisk is correct in this case. Grandstream said that they were just warnings and to ignore them. We have chosen to ignore Grandstream and move to a different phone vendor. regards, Drew PS. I missed your question earlier because it was a reply to an existing thread. If you want to be seen, start a new thread, don't hijack an old one. John Faubion wrote: Am I the *ONLY* one that has this issue? John Faubion -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Faubion Sent: Thursday, November 01, 2007 11:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTP Read too short Hello, I'm getting the following logs: [Nov 1 10:54:37] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short [Nov 1 10:54:39] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short [Nov 1 10:54:40] WARNING[20878]: rtp.c:1138 ast_rtp_read: RTP Read too short Anyone know how to correct this? I'm using SIPConnect from CBeyond and this appears on incoming calls. I haven't had any complaints about voice quality and I haven't seen any dropped calls. Should I be concerned? Thanks, John Faubion -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Testcall
You have other process using at least one of those 1-10 channels. If some other process have it, testcall cannot grab it. Other process could be other testcall instance or Asterisk itslef. On 11/5/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: # ./testcall testcall.conf Chan 1, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025860' to '013331339767' Chan 2, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025861' to '013331339768' Chan 3, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025862' to '013331339769' Chan 4, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025863' to '013331339770' Chan 5, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025864' to '013331339771' Chan 6, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025865' to '013331339772' Chan 7, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025866' to '013331339773' Chan 8, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025867' to '013331339774' Chan 9, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025868' to '013331339775' Chan 10, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025869' to '013331339776' Loading protocol mfcr2 Failed to open channel: Device or resource busy Why??? My testcall.conf is: destination-no 013331339767 protocol-class mfcr2 protocol-variant ar,10,4 protocol-end cpe caller no originating-no 30025860 on-offered accept circuits 1-10 Thanks!!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --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] PRI dialout problem with some numbers...
I have an Asterisk server (1.4.13) using PRI in Monterrey, Mexico. This is really the first server I have used with PRI in Mexico as we normally use MFC/R2. Everything seems to be working except that some numbers always seem to be busy when you dial them. All these numbers belong to different phone companies. I know that with R2 this problem is present if you have a #define DEFAULT_T1 value under 15000 in mfcr2.c (the default used to be 5000). Is there an equivalent value for PRI? The company we are using is Alestra. Here is what I get when we dial a number that belongs to a company called Protel: -- Executing [EMAIL PROTECTED]:1] Set(SIP/199-08be6c00, TIMEOUT(absolute)=3600) in new stack -- Channel will hangup at 2007-11-05 22:03:34 UTC. -- Executing [EMAIL PROTECTED]:2] Dial(SIP/199-08be6c00, Zap/g1/11070665||Ww) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/11070665 -- Zap/1-1 is proceeding passing it to SIP/199-08be6c00 -- Channel 0/1, span 1 got hangup request, cause 31 -- Hungup 'Zap/1-1' [Nov 5 15:03:34] NOTICE[22300]: cdr.c:434 ast_cdr_free: CDR on channel 'Zap/1-1' not posted == Everyone is busy/congested at this time (1:0/0/1) -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Free T1 Card?
http://www.pikatechnologies.com/ -- Kristian Kielhofner Thanks, I guess I wasn't hallucinating! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)
Can anyone please point me in the right direction, provide me with OpenSER configuration, or any pointers on the subject. I tried to read all the material on how to write configuration files for OpenSER, but it is incomprehensible to me, and it is much harder that when I learning Asterisk 3 years ago. Your help is much appreciated. http://www.voice-sistem.ro/downloads/2007.08.29-Admin-Course/von-italy-2007_admin-course.zip LL ___ --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] Free T1 Card?
On Mon, 2007-11-05 at 10:14 -0800, Michael Collins wrote: I recall several months ago that there was a company that was giving away a free 1-port T1 card, with some specific conditions. Do any of you recall who that was? My Google searches are coming up empty and now I’m wondering if I was hallucinating… They sent to me one PIKA inlineMM with 4 FXO ports. Works great. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] OT: Which SIP method to use for thisspecificbehaviour ?
No idea, sorry. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: 05 November 2007 16:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Which SIP method to use for thisspecificbehaviour ? Thanks for the tip. If I may ask, do you if this signaling is support in Asterisk 1.4 ? 2007/11/5, Steve Langstaff [EMAIL PROTECTED] : Search for: Reason: SIP ;cause=200 ;text=Call completed elsewhere From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: 05 November 2007 15:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT: Which SIP method to use for this specificbehaviour ? Hello, Let SIP extensions 1001 and 1002 belong to an Asterisk calling group : whenever an coming call reaches this calling group, both extensions 1001 and 1002 receive a SIP INVITE message which makes these 2 phones starting to ring. When a callee picks up his phone, the other extension receives a CANCEL or BYE message which stops ringing. Is there any option you can include in CANCEL or BYE messages so that the SIP hardphones would understand it shouldn't have to log this call as it has been replied by someone else ? In other words, is there any Alert-info option which can be used to pilot phones call history logs ? I didn't dare to search myself in IETF archives, given the number of standards, SIP is now including. Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free T1 Card?
Is the offer still there? I work at a very poor college would greatly appreciate the ability to get stuff like that. Thanks! Kristian Kielhofner wrote: On Nov 5, 2007 1:14 PM, Michael Collins [EMAIL PROTECTED] wrote: Gang, I recall several months ago that there was a company that was giving away a free 1-port T1 card, with some specific conditions. Do any of you recall who that was? My Google searches are coming up empty and now I'm wondering if I was hallucinating... Thanks, MC http://www.pikatechnologies.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue Statistics reporting
Anyone know of a good package for reporting on Queue statistics from Asterisk? Bob ___ --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] Please explain the correct LED color for B410P
Hi. I have installed B410P in Europe and the cards works more or less ok. My question is what color should the LED's on the back of the card be when connected to the PSTN NT box? Is there anywhere some information on the expected LED color in any given state (idle, call active, cord unplugged etc.)? On my card the lights are shining Red(orange-ish) but flashing to green every now and then and then shining green when there is a call on one of the lines for that port. tnx, Baldvin ___ --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] Two B410P cards in one machine
Hi. I have two B410P ISDN BRI cards in one machine running Asterisk on Ubuntu 7.04. One card connects to the PSTN network and is therefore in TE mode on all four ports and the other card is in NT mode and connects to a PBX. The Asterisk is used to remap features, callerid's and more from the PSTN to the PBX. 1) Is there any special care I need to take regarding the configuration for these cards when they're put together like this? Especially concerning timing between calls bridged from one card to the other (PSTN call comes in, Asterisk answers it and connects to a new call going out on another port to the PBX)? 2) Is there a way to make sure that this is all run on the PSTN timing source through the asterisk box and over to the PBX? 3) Even though the call quality through the Asterisk box is ok as far as I can hear, I'm experiencing tiny drops in the audio stream at regular intervals (around every two seconds or so). My guess was timing slip of some sort between the cards or something like this, but perhaps I'm missing something that really needs to be taken care with when using two cards like this in one machine? Perhaps all the same question with a different twist, but I'm just trying to get the hang of this config and I can't find detailed enough documentation for this scenario via usual sources. All information relating to the correct or proper configuration of multiple B410P cards in one machine is very much appreciated. tnx, Baldvin ___ --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] Kirk IP600/3 Wireless Server SIP config
Where did you buy it , and how much did it cost ? ip600v3, base stations and phones ... 2007/10/26, Benny Amorsen [EMAIL PROTECTED]: RB == Remco Barendse [EMAIL PROTECTED] writes: RB Hi list! Is anyone using the Kirk IP600/3 with SIP firmware RB connected to Asterisk? Yes. RB Any experiences / caveats? Make sure you keep the firmware updated. It improves rapidly. RB If anyone would be willing to share the dump of their IP600 config RB file, i would really appreciate it. Sorry I'm not at work right now. If I get time later, I will. RB Is there anything special i should put in my asterisk config? No, the IP600 is just like any other SIP device. /Benny ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk OpenVZ
Hi All, I've got debian (etch), openvz and asterisk up and running using the openvz wiki guides. The examples use `apt-get install asterisk` and this will install 1.2.13. Has anyone gotten an VPS to compile the latest versions from source? Also, I'm unsure how the zaptel modules come into play, could use some guidance there as well. Thanks. JR -- JR Richardson Engineering for the Masses ___ --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] 1.4 SIP Jitter Buffer
Hello, I'm running into a few situations on lossy network links where a SIP jitter buffer w/ some PLC would be helpful. My main TDM gateways are running 1.2 (which is solid, stable, reliable and very very very well behaved when you know it's limitations), but I'm considering upgrading them before the end of the year to 1.4. Two of the main reasons that I would do this are Variable Length DTMF and SIP Jitter Buffering. I would be very interested in hearing from anyone that is actually running 1.4 in a PRODUCTION environment, gatewaying SIP to TDM using Digium cards. To me, production means being able to have 3-4 PRI circuits maxed out for 12+ hours a day and 7+ call setups / second. Anything less than that is not really going to be an accurate comparison to what I have running. Anyone have any feedback about this combination? ___ --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] 1.4 SIP Jitter Buffer
Gregory We have many Asterisk 1.4.13 in production solid like a rock. Couples examples: a) Asterisk 1.4.13 + Unicall + 2 E1 MFCR2 Digium + Legacy PBX 60+ Extentions / IVR / 10~30 concorrent calls b) Asterisk 1.4.11 + 1 E1 ISDN PRI Digium 50+ Extentions / IVR / 5 Queues / ~2000 call/day c) Asterisk 1.4.13 + 4 E1 ISDN Digium (working in progress) CallCenter / 150 PAs / 15 Queues / expected 8000 calls/day -- Luc Gregory Boehnlein escreveu: Hello, I'm running into a few situations on lossy network links where a SIP jitter buffer w/ some PLC would be helpful. My main TDM gateways are running 1.2 (which is solid, stable, reliable and very very very well behaved when you know it's limitations), but I'm considering upgrading them before the end of the year to 1.4. Two of the main reasons that I would do this are Variable Length DTMF and SIP Jitter Buffering. I would be very interested in hearing from anyone that is actually running 1.4 in a PRODUCTION environment, gatewaying SIP to TDM using Digium cards. To me, production means being able to have 3-4 PRI circuits maxed out for 12+ hours a day and 7+ call setups / second. Anything less than that is not really going to be an accurate comparison to what I have running. Anyone have any feedback about this combination? ___ --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
