[asterisk-users] ISDN BRI vs SIP Trunks over EDIA
Hello, I am installing Asterisk in a small office with just 4 lines and 8 Extensions. I have two choices from my local telco (Fairpoint): 1) Old School ISDN BRI lines which I would connect to Asterisk with a OpenVOX B200P 2) Telco supplied SIP trunks over a service called EDIA which is 1MB ethernet over several pair of copper lines. The ISDN BRI solution is less than 1/2 the price of the SIP solution. Any recommendations? Pitfalls? Mark Engelhardt - in snowy Vermont! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Testing 911 call
Joseph, I have made a quite a few test calls to 911. They don't charge you and they don't get upset. Just let them know right away it is a non-emergency test call, and then let them know who you are and what you need to verify on their information screen. Mark Engelhardt On May 5, 2013, at 11:07 AM, Joseph wrote: How to test 911 call? I'm using Audiocodes and it setup to strip the first number but I've never tested the 911 call. I don't want to go live as they might charge me. -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Noise on phones while speaking...
Carlos, I think the noise you are hearing might echo cancelation that is broken or set incorrectly. Maybe the card and asterisk are both trying to echo cancel? Mark On Nov 13, 2012, at 1:52 PM, Carlos Chavez wrote: I have a new install and the customer is complaining that they hear noise on all calls, no matter if it is internal or external, desk phones or softphones. The noise is only present when the user is speaking, not the remote side. The remote side does not hear the noise, only the local user. We are using Asterisk .1.8.11-cert8 on a CentOS 6 machine with a Digium AEX800 card and DAHDI 2.6.1. I really do not know how this noise is generated. Where can I look? Why would a SIP to SIP call have this noise? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set(CALLERID(name)) when incoming call is anonymous
Kevin,
You might have luck changing the callerid number so its not empty, that might
override the Anonymous label.
exten = 12345678,1,GotoIf,$[${LEN(${CALLERID(num)})} != 0]?3
exten = 12345678,2,Set(CALLERID(num)=0)
exten = 12345678,3, Your code starts here
Good luck!
Mark
On Feb 16, 2012, at 9:26 PM, Kevin Shanahan wrote:
Hi,
I'm trying to figure out why I can't pass through caller ID details
that I set manually if the incoming call that I am forwarding was
anonymous.
Our reception staff need to know which number the client was calling
in on so they can give the right greeting message when answering.
E.g. I have the following in our dialplan for one reception number
(similar for others):
G_RECEPTION=SIP/SIP/
exten = 12345678,1,Set(CALLERID(name)=ORG1)
exten = 12345678,n,Set(CALLERID(name-pres)=allowed)
exten = 12345678,n,Dial(${G_RECEPTION},15,i)
exten = 12345678,n,VoiceMail(12345678,su)
exten = 12345678,n,Hangup()
Normally this works great with the name ORG1 and the client's number
both appearing on the handset (Snom 320). However, if the caller had
no caller ID this shows up on the screen as Anonymous.
Setting name-pres actually doesn't have noticable effect. I added that
later when trying to find a solution.
How can I make ORG1 show up on the screen when the caller has no
caller ID?
TIA,
Kevin Shanahan
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Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
Bilal, I suggest you turn on logging on your tftp server to see what files are actually being requested, and if the the tftp server is dishing them out... Try adding a few v's to your tftp setup: File: /etc/xinetd.d/tftp Line to change: server_args = -s /tftpboot -v -v -v Look in /var/log/messages for the output. Also, I believe your 7942G has a console/aux port which is a serial port, you can learn what is happening as the phone boots up with that too. Good Luck! Mark On Jun 13, 2011, at 3:02 PM, bilal ghayyad wrote: Dears; The Asterisk version is 1.8.3.2 The Cisco IP Phone is 7942G and it is running now skinny. The used TFTP is tftp-server which is installed in fedora. I placed the following two files (but look like it was not taken from the TFTP, as nothing appeared in the messages), but I am able to to ping from the asterisk box to the vlan that the Phone is connected, so no problem in the reachability: SEPB8BEBF22AB62.cnf.xml xmlDefault.CNF.XML Are the files name correct? Or the Cisco IP Phone 7942G are not working fine with Asterisk or the tftp-server? Regards Bilal Hi All; Can anyone advise if using Cisco IP Phones Which model(s) are you planning to use ? in skinny protocol is fine or not? Or it is better to use it in SIP protocol? -- Hi, On 06/13/2011 01:04 PM, bilal ghayyad wrote: Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? SCCP works better than SIP in my opinion as there are more features. Check out http://chan-sccp-b.sourceforge.net/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant.
Hello Everyone. We have: Asterisk 1.4.21.2 zaptel-1.4.11 libpri-1.4.5 Sangoma A101D Connected to a PRI Cicso 7960G phones (About 30 of them) We have a problem with dropped calls that has going on for a long time. We get up to 5 dropped calls on a bad day. They all seem to be incoming calls. I have a recording of what my users report a dropped call sounds like right before it drops http://www.stepawayfromthecomputer.com/drop.wav Please have a listen to the recording and tell me what you think it means I am looking for any ideas as to what I should do to track this down. I would love a lead on a good consultant who can help fix this. Mark ___ -- 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] Dropped Call Problem -- Looking for ideas and a consultant.
Conner, I contacted my telco and they report they have a: EWSD Siemens Central Office Which does not support 5ess Any other way around this? How did you determine changing to 5ess would fix your problem? Mark On Jul 10, 2009, at 12:15 PM, Connor Spiess wrote: We had the same problem using a Digium T1 card. We switched the coding to from NI2 to 5ess and we haven't dropped a call since. You will have to check with your service provider to see if they do 5ess. Connor Spiess Network Specialist -Original Message- From: Mark Engelhardt [mailto:ma...@intuitiveengineering.com] Sent: Friday, July 10, 2009 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant. Hello Everyone. We have: Asterisk 1.4.21.2 zaptel-1.4.11 libpri-1.4.5 Sangoma A101D Connected to a PRI Cicso 7960G phones (About 30 of them) We have a problem with dropped calls that has going on for a long time. We get up to 5 dropped calls on a bad day. They all seem to be incoming calls. I have a recording of what my users report a dropped call sounds like right before it drops http://www.stepawayfromthecomputer.com/drop.wav Please have a listen to the recording and tell me what you think it means I am looking for any ideas as to what I should do to track this down. I would love a lead on a good consultant who can help fix this. Mark ___ -- 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 Mail was checked for spam by the Freeware Edition of No Spam Today! The Freeware Edition is free for personal and non-commercial use. You can remove this notice by purchasing a full license! ___ -- 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] Dropped Call Problem -- Looking for ideas and a consultant.
Steve, Thanks for your thoughts. I am tearing out my last bit of hair on this one. We only use sip on our internal network to talk to the 7960s We are getting drops from no-cell phone hard wired phones too. Unfortunately There are too many drops for me to let this go. :( Mark On Jul 10, 2009, at 12:48 PM, Steve Totaro wrote: This is an age old Asterisk (and general telephony) problem. I can't blame it all on Asterisk. Never thought of the 5ess, filed in my memory bank as this is an age old problem. Too bad it happens with SIP providers and not just the little guys but XO for instance. I hear crackling. Cell phones drop all the time. On a bad day I get five dropped cell phone calls a day. Thanks, Steve Totaro On Fri, Jul 10, 2009 at 12:15 PM, Connor Spiess cspi...@idea- ma.com wrote: We had the same problem using a Digium T1 card. We switched the coding to from NI2 to 5ess and we haven't dropped a call since. You will have to check with your service provider to see if they do 5ess. Connor Spiess Network Specialist -Original Message- From: Mark Engelhardt [mailto:ma...@intuitiveengineering.com] Sent: Friday, July 10, 2009 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant. Hello Everyone. We have: Asterisk 1.4.21.2 zaptel-1.4.11 libpri-1.4.5 Sangoma A101D Connected to a PRI Cicso 7960G phones (About 30 of them) We have a problem with dropped calls that has going on for a long time. We get up to 5 dropped calls on a bad day. They all seem to be incoming calls. I have a recording of what my users report a dropped call sounds like right before it drops http://www.stepawayfromthecomputer.com/drop.wav Please have a listen to the recording and tell me what you think it means I am looking for any ideas as to what I should do to track this down. I would love a lead on a good consultant who can help fix this. Mark ___ -- 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 Mail was checked for spam by the Freeware Edition of No Spam Today! The Freeware Edition is free for personal and non-commercial use. You can remove this notice by purchasing a full license! ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Dropped Call Problem -- Looking for ideas and a consultant.
More info on my dropped call issue: Here is a report on a dropped call from today: Call Started echoing then cut out Stats From the 7960 Stats Screen: RxCnt:011853 TxCnt:010204 MaxJtr: 762 RxLost: So, now I am starting to suspect that I have this problem: http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-ce-2-4-2-cisco-7960-7940-receiving-audio-drop-during-c Any suggestions? Mark On Jul 10, 2009, at 2:45 PM, Steve Totaro wrote: DSS? Ask them what other signaling they can support. I would escalate every day if I were you. It is the only way to get things done. Get it to the top and be mad, even if you are not. When fixed, PRAISE everyone from top to bottom. A level one tech will say Ah you are using Asterisk, we don't support that, or Sorry, our switch cannot do 5ESS (when it can but takes a bit of work). Thanks, Steve Totaro On Fri, Jul 10, 2009 at 2:24 PM, Mark Engelhardt ma...@intuitiveengineering.com wrote: Conner, I contacted my telco and they report they have a: EWSD Siemens Central Office Which does not support 5ess Any other way around this? How did you determine changing to 5ess would fix your problem? Mark On Jul 10, 2009, at 12:15 PM, Connor Spiess wrote: We had the same problem using a Digium T1 card. We switched the coding to from NI2 to 5ess and we haven't dropped a call since. You will have to check with your service provider to see if they do 5ess. Connor Spiess Network Specialist -Original Message- From: Mark Engelhardt [mailto:ma...@intuitiveengineering.com] Sent: Friday, July 10, 2009 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropped Call Problem -- Looking for ideas and a consultant. Hello Everyone. We have: Asterisk 1.4.21.2 zaptel-1.4.11 libpri-1.4.5 Sangoma A101D Connected to a PRI Cicso 7960G phones (About 30 of them) We have a problem with dropped calls that has going on for a long time. We get up to 5 dropped calls on a bad day. They all seem to be incoming calls. I have a recording of what my users report a dropped call sounds like right before it drops http://www.stepawayfromthecomputer.com/drop.wav Please have a listen to the recording and tell me what you think it means I am looking for any ideas as to what I should do to track this down. I would love a lead on a good consultant who can help fix this. Mark ___ -- 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 Mail was checked for spam by the Freeware Edition of No Spam Today! The Freeware Edition is free for personal and non-commercial use. You can remove this notice by purchasing a full license! ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Debug dropped calls
Hello, I have nearly the same issue. Does anyone have a suggestion as to how to find and fix this problem? Mark On Jul 16, 2008, at 10:59 AM, Mike (Asterisk) wrote: [zaptel] span=1,0,0,esf,b8zs At least one of your spans should be getting it's timing from your service provider. It looks like that would be span one, this should read: span=1,1,0,esf,b8zs I checked my config files from before my upgrade, and I do have span 1 setup as you indicated. Silly oversight on my part. I will make the change and restart Asterisk/zaptel when I can. However, I experienced dropped calls before the upgrade, but I'll make this change and go from there. [zapata] faxdetect=incoming In the past, faxdetect has been known to cause problems. I'll change zaptel.conf first so I only change one thing at a time. I was finally able to change my configuration for span 1 from my provider to by primary clock source and span 2 to be a secondary source. Aside from the configuration file, I'm unable to locate a way from within asterisk to confirm that the change took effect. 'pri show span 1' shows: (2 shows the same, just the D channel as 48) Primary D-channel: 24 Status: Provisioned, Up, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 Is there some text in pri intense debug that would confirm the clock source? -Mike - The information contained in this e-mail message is confidential and/ or privileged and is intended only for the use of the individual or entity named above. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. If you are not the intended recipient you are hereby notified that any unauthorized disclosure, copying, distributing or taking any action in reliance of contents of this information is strictly prohibited. - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help Needed: Can't make local calls on a brand new PRI.
Hello, I just installed a PRI and when I make a local (seven digit) call, I get Code 28 back from the telco, (I believe code 28 means Invalid Number) and I hear a fast busy on the phone. Here is the output: -- Executing Dial(SIP/marke-17b1, ZAP/G1/4967171) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G1/4967171 -- Zap/23-1 is proceeding passing it to SIP/marke-17b1 -- PROGRESS with cause code 28 received -- Zap/23-1 is making progress passing it to SIP/marke-17b1 As you can see, asterisk is reporting 4967171 as the dialed number (which is valid) When I dial long distance, everything works fine. Here is the output from long distance... -- Executing Set(SIP/marke-80f8, CALLERID(all)=Small Dog Electronics8005116277) in new stack -- Executing Dial(SIP/marke-80f8, ZAP/G1/17077510895) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G1/17077510895 -- Zap/23-1 is proceeding passing it to SIP/marke-80f8 -- Zap/23-1 is ringing If I just send the full 18024967171 to the telco, I get a voice from the telco saying it is not necessary to dial 1 or the area code when calling this number. So the questions: Is there anyway to further verify that asterisk is not sending any extra digits or filler digits to the telco on the PRI? If the problem is not in asterisk or zaptel, what do I say to the Telco to get them to believe the problem is on their end? We are running: Asterisk 1.2.6 Zaptel 1.2.14 TE110P Card Mark Engelhardt ___ --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] Help Needed: Can't make local calls on a brand new PRI
Hello, I just installed a PRI and when I make a local (seven digit) call, I get Code 28 back from the telco, (I believe code 28 means Invalid Number) and I hear a fast busy on the phone. Here is the output: -- Executing Dial(SIP/marke-17b1, ZAP/G1/4967171) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G1/4967171 -- Zap/23-1 is proceeding passing it to SIP/marke-17b1 -- PROGRESS with cause code 28 received -- Zap/23-1 is making progress passing it to SIP/marke-17b1 As you can see, asterisk is reporting 4967171 as the dialed number (which is valid) When I dial long distance, everything works fine. Here is the output from long distance... -- Executing Set(SIP/marke-80f8, CALLERID(all)=Small Dog Electronics8005116277) in new stack -- Executing Dial(SIP/marke-80f8, ZAP/G1/17077510895) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G1/17077510895 -- Zap/23-1 is proceeding passing it to SIP/marke-80f8 -- Zap/23-1 is ringing If I just send the full 18024967171 to the telco, I get a voice from the telco saying it is not necessary to dial 1 or the area code when calling this number. So the questions: Is there anyway to further verify that asterisk is not sending any extra digits or filler digits to the telco on the PRI? If the problem is not in asterisk or zaptel, what do I say to the Telco to get them to believe the problem is on their end? We are running: Asterisk 1.2.6 Zaptel 1.2.14 TE110P Card Mark Engelhardt ___ --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] Help Needed: Can't make local calls on a brand new PRI
That has been the setting all along. Matt and I will try pri intense debug span 1 in the morning and we can go from there. Thanks so much for your help so far. I am also trying to get the telco to tell us what they are actually seeing on their side, The report of 000 at the end of the number was not really confirmed when I quizzed them today. Mark On Feb 28, 2007, at 9:45 PM, Steve Totaro wrote: Matt wrote: It is currently set to unknown. switchtype=national signalling=pri_cpe pridialplan=unknown Was it originally or did you just change it? Did you stop Asterisk and do ztcfg after making the changes? Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T1/E1 Multiflex Voice/WAN Interface Card VWIC-1MFT-T1 connection to asterisk Advice Needed.
Hello, I am considering deploying a asterisk system using a VWIC-1MFT-T1 installed in a cisco router. here is my basic plan: Telco PRI/T1 --- cisco 2600 router (with a VWIC-1MFT-T1 card) -- asterisk server --- 30 cisco 7960 phones I have some questions before I spend the $ on this plan. How well does this sort of installation work? Will I get caller ID? Any shortcomings to this idea? Can I connect two asterisk servers to this device? If I decide to do this, I would be looking for consulting talent to help with the initial configuration, Contact me off list if you can help with that. Mark Engelhardt ___ --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] Cisco 7960 password/shared secret problem --- Related to OS X ?
Hello, Whenever I put in a password/Shared Secret in my 7960 and try and get it to register with asterisk on OS X setup, the phone fails to register. Oct 31 20:03:46 NOTICE[989]: chan_sip.c:11045 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '67.121.71.120' - Wrong password When I make the password blank in both sip.conf and SIP.cnf then it registers and works perfectly. The same phone connects with a linux based [EMAIL PROTECTED] installation without any problems. 3 Questions: 1) How can I tell the actual password/shared secret the phone is transmitting to Asterisk. 2) Has anyone had any trouble configuring 7960s with passwords via the SIP.cnf file? 3) Do you think there is a bug in Asterisk that comes out when its compiled for PowerPC OS X ? Please let me know if you have any suggestions too. Mark Engelhardt Here is my setup: I have a Cisco 7960s running sip image : P003-08-4-00 I am setting the passwords via tftp server and the SIPmacaddress.cnf file. Line 4 on the phone: Asterisk on OS X : Asterisk 1.2.10 running on Mac OS X Server 10.4 (tiger) The portion of the .cnf file: line4_name: 575 line4_authname: 575 line4_password: line4_displayname: 575 line4_shortname: 575 Line 1 on the phone: Asterisk on Linx Asterisk 1.2.5 running on Linux line1_name: 701 line1_authname: 701 line1_password: password line1_displayname: 701 line1_shortname: ___ --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
