RE: [asterisk-users] How to detect long calls
You should: Set(TIMEOUT(absolute)=14400) When the call is received - this will set the maximum limit of a call and asterisk will force hang-up when the limit is reached. 14400 seconds = 4 hours, which for our purposes is longer then any call we expect. Even if you double-it or set it to several days some limit is better then nothing. When we found the same problem we had a call that was stuck open for 20 days. The call was stuck in a conference and was sending the on-hold music, which is what kept it open. Hope that helps. Cullin J. Wible -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: Tuesday, January 16, 2007 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to detect long calls Savoy, Kevin - Williston, ND wrote: We have been running an Asterisk box with 1.2.9.1 on it since August in a call center environment. We use the Asterisk box as an IVR and then pass the calls on to a Nortel Option 11C. Today we found in our long distance bill two calls that lasted a VERY long time. One was 58 hours and another was 38 DAYS!!! Nortel does not show this call being that long. Obviously the person that called in didn't hold the line for 58 days so somehow between Asterisk and MCI the call got stuck open and didn't hang up on the network. My question is two parts, part one, has anyone heard of anything like this where a call doesn't hang up properly and seems stuck in the system. Part two is there anyway to monitor in Asterisk the length of all active calls and then if a call lasts longer then, say one hour, we could send off a text message or warning. Hi , similiar thing happend to me. Try looking at the L() optin in Dial. I define a max call time, say few hours, then warn every x seconds, then cut the call. -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Audiocodes GPL
There is nothing in the GPL that prohibits you from selling the software (RedHat Software). There is also nothing stops a sales person from denying it. They must provide a copy of the GPL and they must give you the source code and related modifications if you ask (not sure if you have). There are other terms and depending on who you ask, lots of interpretation. If you truly believe there is a violation (which I doubt) you should contact the Free Software Foundation - they wrote the license. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Tuesday, January 16, 2007 6:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Audiocodes GPL Am Mittwoch, den 17.01.2007, 07:38 +0800 schrieb Leo Ann Boon: Andrew Joakimsen wrote: I have some Audiocodes units which appear to be running Linux, according to the unit's own System Log kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006 Googling turns up: http://www.jungo.com/openrg/openrg.html OpenRG is a Linux based device platform. So, Audiocodes probably licensed it from Jungo. Just because the unit runs Linux, doesn't necessarily imply that there's a GPL violation. Surely not. Linux is intented to be used in proprietary hardware, applications et cetera. But, if I am not mistaken, if a device uses any GPL'd software, this must be clearly stated by the vendor, a copy of the GPL must be handed along with the device and you have the right to obtain a copy of all open source source code files involved in the project, for a marginal charge. Outright denial of the usage of Linux in such a device seems to not comply with that. If you intend to pursue this, you could try to find information on www.gpl-violations.org (and no, this is not an organisation that helps to violate the GPL ;-) BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] same extension on softphones and hardphones
This is not a SIP issue, but a problem with your configuration.
We have all hard phones register/authenticate with their MAC address as the
user/peer name. Soft phones use user id's that correspond to the person. We
then have our dialplan ring the appropriate devices (soft or hard) depending
on which extension was dialed.
Use the operator in the dial string to ring multiple devices.
Cheers,
Cullin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeronimo
Romero
Sent: Sunday, November 12, 2006 6:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] same extension on softphones and hardphones
Is this inherently an issue with sip? Is it possible for a sip server to
actually ring two different sip registration from the same account or is
this not possible under any sip enabled pbx?
Thanks again
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin
Hoffmeister
Sent: Sunday, November 12, 2006 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] same extension on softphones and hardphones
Am Sonntag, den 12.11.2006, 16:29 -0500 schrieb Jeronimo Romero:
Sorry if you see this message repeated twice. I would like to set up
hard phones and softphones with the same extension so that when
anybody
in the company dials an extension, each user's hardphone and softphone
will ring at the same time. I've tried setting this up before, but I
noticed that the last sip device to register with the same extension
is
the only one that rings when the extension is dialed. The sip devices
they will be using are Grandstream GXP2000 desktop phones and Xten
Eyebeam softphones. Each user will have one of each. What is the best
way to accomplish this?
Xten eyebeam ext 110 \
\
-- Asterisk 1.2.8
/
GXP2000 phone ext 110 /
One possible solution is to have one sip account for each _device_, not
extension; say sip110h and sip110s for the 110-user.
Then use the dial command in your extensions.conf like
exten = _1XX,1,Dial(SIP/sip${EXTEN}sSIP/sip${EXTEN}h)
This will cause parallel ringing phones. First come first serve.
Hth
Anselm
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RE: [asterisk-users] Increase VoiceMail Messages Recording Gain -AudioCalls are Ok
externnotify=/opt/asterisk-1.2.7.1/sbin/vm-audio vm-audio uses 'sox -e' to determine how much to scale by without clipping and then Then 'sox -v' to scale the sound file. This happens after the email message is sent, but by changing the order of a few lines in the app_voicemail.c program you can have the externnotify run before the email message is sent. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Wednesday, October 11, 2006 12:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Increase VoiceMail Messages Recording Gain -AudioCalls are Ok That doesn't always work :) There's two options... either port the volgain patch from 1.4 to 1.2 (If anyone wants a copy, we've been using it for months... however it also converts to mp3 so we'd have to strip that out)... or use 1.4 which includes the patch. Let me know if I should post a copy of the older code somewhere. The 1.4 patch is here: http://bugs.digium.com/view.php?id=6237 Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 On Tue, 2006-10-10 at 21:18 -0500, Shawn Kelley wrote: I had the same problem. Checking voicemail via the phone was perfectly normal but the email attachments were so quiet we had to turn the computer volume all the way up along with the speakers amps just to make the attachment understandable. Then just wait until someone forgets to turn the volume back down and a lovely windows message box pops up. Scares the (pick your word) out of everyone in the office! After much searching I found the solution: In the voicemail.conf file change the order in which the recording formats are specified. Asterisk will email the first format in the list. My original line: format=wav49|wav|gsm My new line: format=wav|wav49|gsm NOTE: My understanding is that the wav files are much larger attachments than the wav49 version. However, we haven't noticed much difference, still fairly small attachments. Definitely no problems on a LAN or Broadband connection. From: Marco Mouta [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 10, 2006 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok Hi all I'm deploying a VoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions. On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application. I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX are just fine, at least no complaining from users, seems good to me:) The problem is: Right now, and i'm referring only to calls directly handled by VoiceMail application, the users get their audio files in email but the audio is very very low. I've thought about changing RX gain on PRI interface between legacy pbx and asterisk, but until now no complaining with audio calls. I'm afraid that changing this parameter to solve voicemail issues will get me in troubles with Voice Calls . Any advice, or previous similar experience? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Understanding NAT Traversal
1) The difference between a web browser using port 80 and SIP on port 5060 is as follows: - HTTP uses TCP, which maintains state and therfore can be tracked in a NAT table. SIP uses UDP, which is a stateless protocol, which is more difficult to track. (This by itself isn't a big deal, since UDP+NAT work well together). - The source port for a web browser is random, the destination port is 80. So you have an random-80 request and 80-random response. In SIP you have a 5060-5060 request and a 5060-5060 response. It is very difficult to track a a many-to-one NAT (technically port address translation (PAT)) when you can't change the source or destination ports. For those who have ever had problems NAT'ing GRE/IPSec VPN's this is the same issue. 2) Responses to new port numbers to a NAT'ed host don't work without special code on the NAT'ing box. Linux has code to support this for Real Audio, Quake, FTP, and others. Perhaps someone needs to write the iptables_sip_helper module. 3) The Upnp network device would be the smame as #2, except that UPNP doesn't do this type of thing so it's totally irrevelant here. 4) STUN Can help with discovering the external address, but this, combined with a fixed port PAT is what really causes the underlying issues with SIP NAT traversal. Additionally, due to the X-OR checksums that are done with STUN it will only work through 1-level of NAT. 1 Levels of NAT will cause STUN to determine that its IP address has immediately changed and to re-fresh and re-register as soon as possible (which could easily bring down your server, as we have seen). Conclusions: If you can eliminate 1/2 of the NAT issue (run asterisk or a SIP proxy on a public address) you will be able to solve all of your issues. This combined whith a few settings such as responding to the report port (instaed of 5060), etc (asterisk standard NAT settigns) will do all that you need. After spending lots of time with all of this: If you're running STUN you're trying too hard. Cullin J. Wible -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hugolivude Sent: Tuesday, October 10, 2006 9:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Understanding NAT Traversal Thanks Moj! The RTP packet problem makes sense. Still unclear on some of the other points: I think the biggest problem with SIP is the RTP ports. The initial SIP request goes out (for example) to port 5060, and FROM port 5060 as well. The response needs to get back to the SIP UA on that port (which would limit the nat router to only be able to deal with ONE internal ua at a time, if they were both using the standard port 5060), which could conceivably happen easily enough. An Internet browser uses port 80. I might have two or more behind a NAT both using port 80. Isn't that the same thing? But in the SIP handshake more ports are chosen, typically in the 10,000 to 20,000 range. The NAT router would then need to be configured to direct that anticipated RTP stream to the proper internal client. That just doesn't happen :) Hmmm, that makes sense. For various reasons, I'm not too partial to UPnP, but maybe there needs to be a SIP UA that uses UPnP to configure a NAT router for it, when an RTP stream is begun? Not following this part... Now the clincher to all of this is that I'm merely talking about the ip packets transferred and their return addresses. While I'm not qualified or experienced enough to comment on problems that might arise with the contents of the SIP headers themselves, I suspect that's where the REAL trouble lies with SIP NAT traversal. The SIP UA needs to put the proper return address in the SIP headers before the lower layers of the OSI model take over. It can't know its externally-visible ip address unless A) the user manually enters it or B) it can ask some outside server what it's perceived address is. Isn't this what a STUN server does? Sends an HTTP message to SIP UA so that the SIP UA can strip out the external IP address of the NAT? Thanks again, H ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.2.10 - g726 Issues
Yeah, that's exactly the problem that I am having here (also switched to g729 and gsm). However, Teliax has told me that the g726 issue is with the * 1.2.10 release and as a result not an issue with their service. I'm not entirely convinced, but since we also use g726 for some of our internal phones we must support it and if it's broken in 1.2.10 then I won't upgrade. What version of * are you runing? Thanks, Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Tuesday, August 15, 2006 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.2.10 - g726 Issues Cullin J. Wible wrote: I have hard that 1.2.10 has issues with voice quality through g726. Can anyone provide any feedback or point me in the right direction about the current status of this problem? Been using g726 between multiple * systems for some time and the quality has been very good. Recently, however, all calls via teliax.com using g726 have had very poor quality. Switching back to gsm for them cleared up the iax audio nicely. Not sure if teliax changed something or what, but had been working fine for several months. R. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.2.10 - g726 Issues
Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, August 16, 2006 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.2.10 - g726 Issues I'm running SVN-trunk-r16869M (compiled 2006-04-01) with iax trunks to five remote systems most of which are v1.2.10. No problems with any of those trunks using g726. Teliax is the only system that I've had any issues with using iax and g726. I've not tried sip to them and don't have any intentions of doing that right now. R. Cullin J. Wible wrote: Yeah, that's exactly the problem that I am having here (also switched to g729 and gsm). However, Teliax has told me that the g726 issue is with the * 1.2.10 release and as a result not an issue with their service. I'm not entirely convinced, but since we also use g726 for some of our internal phones we must support it and if it's broken in 1.2.10 then I won't upgrade. What version of * are you runing? Thanks, Cullin -Original Message- Cullin J. Wible wrote: I have hard that 1.2.10 has issues with voice quality through g726. Can anyone provide any feedback or point me in the right direction about the current status of this problem? Been using g726 between multiple * systems for some time and the quality has been very good. Recently, however, all calls via teliax.com using g726 have had very poor quality. Switching back to gsm for them cleared up the iax audio nicely. Not sure if teliax changed something or what, but had been working fine for several months. R. ___ --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] 1.2.10 - g726 Issues
I have hard that 1.2.10 has issues with voice quality through g726. Can anyone provide any feedback or point me in the right direction about the current status of this problem? Thanks, Cullin ___ --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] Voicemail volume adjustment
Good catch - I hadn't realized that. You are correct that in app_voicemail.c sendmail is run prior to the externnotify script. I see a few options: 1) change the code in app_voicemail.c 2) Use the externotify script to assemble and send the email messages 3) Run a web server and include a link to the voicemail message instead of attaching it. None of them look fun. Not sure how many * developers read this list, but it would be great of the run_externnotify(vmu-context, vmu-mailbox); call in notify_new_message() in app_voicemail.c could be moved to the top of the function as it is probably the preferred solution. Cullin J. Wible Co-Founder CTO Email Data Source, Inc. 212-514-8900 x1006 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins Sent: Wednesday, June 28, 2006 8:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Voicemail volume adjustment This works great, however, when I look at the full log, it says that the sendmail is executing prior to vm-audio. Any way to change this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cullin J. Wible Sent: Tuesday, June 27, 2006 8:41 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Voicemail volume adjustment In voicemail.conf: externnotify=/opt/asterisk-1.2.7.1/sbin/vm-audio The attached script should increase as much as possible without clipping. Cheers, Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Technical Support Sent: Tuesday, June 27, 2006 3:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail volume adjustment I frequently find voice messages are emailed to users with insufficient volume - barely audible. I would like to have asterisk run a sox command to adjust the volume of each message before emailing (perhaps once the message has been left). Has anyone done this? Care to share the steps? Thanks, MD ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP - ConfCalling
We run them with 1 call per line, but when we first set them up they would do 8. The problem was switching between calls on a single line. At that time, however, the phone did not return busy and allowed the calls to stack up. This is set in the XML configuration files. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Wednesday, June 28, 2006 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP - ConfCalling Do you have more than one call per line enabled on the Poly? Is it the phone or asterisk returning the busy? What does the console say? On Jun 27, 2006, at 5:29 PM, Mike Staver wrote: I have one extension setup for each Polycom 501 I have, and when I try to call out on a conference call, I get all circuits busy for the second call. I have one sip trunk set up for each DID that I have through our VoIP provider. Each trunk is capable of having one call placed on it at one time. So, I'm thinking I need a way to tell Asterisk to have the second call go out on one of the other empty trunks at the time if one exists, which more than likely, it will. Is this possible? -- -Mike Staver [EMAIL PROTECTED] [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Remote employees using Polycom 501 lose abilityto receive incoming calls after few minutes.
Polycom phones support STUN - that should solve the issue too. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dr. Michael J. Chudobiak Sent: Wednesday, June 28, 2006 1:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Remote employees using Polycom 501 lose abilityto receive incoming calls after few minutes. Von L. wrote: plugged in. They work immediately after being plugged in, but they lose the ability shortly thereafter. They can always make outbound calls, but only to real phone numbers, not extensions. They each have NAT routers, and I have triple checked that they have opened/forwarded the correct ports, basically 5060-3 UDP. Once they See the NAT Issues section at http://www.voip-info.org/wiki/view/IAX. (The page is for IAX2, but the NAT issues are relevant for UDP SIP ports too). Basically, some NAT routers forget UDP mappings after a VERY short time (like 30 seconds). Took me a while to figure that out. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail volume adjustment
Because: usg(2)[EMAIL PROTECTED] causes the app to exit with a non-zero status, Because: [EMAIL PROTECTED]|g(2) causes asterisk to hard crash. And trying to use g2 in either case doesn't work either. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dustin Wildes Sent: Wednesday, June 28, 2006 1:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail volume adjustment Why use an application like sox - when you can make the voicemail application do it natively: exten = s,1,Dial(SIP/100,10) exten = s,2,Voicemail([EMAIL PROTECTED]|ug(10)) The key is the g(10) parameter: From the 'show application voicemail': g(#) - Use the specified amount of gain when recording the voicemail message. The units are whole-number decibels (dB). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Technical Support Sent: Tuesday, June 27, 2006 3:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail volume adjustment I frequently find voice messages are emailed to users with insufficient volume - barely audible. I would like to have asterisk run a sox command to adjust the volume of each message before emailing (perhaps once the message has been left). Has anyone done this? Care to share the steps? Thanks, MD ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Modifying Voicemail menus?
We did it by comment out a number of lines in the code and then re-compiled just that module. We also did the same for the company directory. Other then that I'm not sure if there's much you can do. Cullin J. Wible Co-Founder CTO Email Data Source, Inc. 212-514-8900 x1006 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Elder Sent: Tuesday, June 27, 2006 12:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Modifying Voicemail menus? Is there a way to edit the options available in the voicemail menu trees? My users are complaining that it's too complicated (I know, it's not really complicated), and I wanted to remove some of the options if this is possible. So far I havent' found any info on the wiki or searches, not that it isn't out there.. I just cant' seem to find it.. Any pointers? Thanks Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail volume adjustment
In voicemail.conf: externnotify=/opt/asterisk-1.2.7.1/sbin/vm-audio The attached script should increase as much as possible without clipping. Cheers, Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Technical Support Sent: Tuesday, June 27, 2006 3:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail volume adjustment I frequently find voice messages are emailed to users with insufficient volume - barely audible. I would like to have asterisk run a sox command to adjust the volume of each message before emailing (perhaps once the message has been left). Has anyone done this? Care to share the steps? Thanks, MD ___ --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 vm-audio Description: Binary data ___ --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] best hardphone for Asterisk?
We've used a number of the polycom 301 and 501 phones in our office. We have also deployed a dozen of the Linksys SPA-1001 single-line FXS adapters using G726, SIP, NAT and STUN. They are extremely reliable and easy to deploy - $60-$70 US each. We tested a number of IAX hard phones and didn't find anything that was reliable and/or suitable for our corporate setting. We really wanted to run IAX for remote users, but eventually decided that SIP/STUN was easier to support. We also tested the IAXy device and found that it's inability to use DNS resolution, only be configured on Linux, and only run ulaw/alaw made and that it cost more then the SPA-1001, which can use DNS, G726/G729 and has web-based configuration for less money the more attractive option. We also tested the IAX hard phone made by AT-COM only to find that a number of features such as call transfer do not work. For home/remote users: setup STUN, and use a SPA-1001. For a corporate setting I highly recommend the Polycom phones. Cheers, Cullin J. Wible Co-Founder CTO Email Data Source, Inc. 212-514-8900 x1006 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Crompton Sent: Monday, June 26, 2006 11:49 PM To: Iain Barker Cc: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] best hardphone for Asterisk? Iain, Thanks for the repsonse but you are kidding me right? From what I can see if I bought this phone and two remotes my outlay would be close to $800 US. This is NOT a home device unless you have nothing better to do with your money! You can buy a lot of single line wireless phones and FXS devices for that amount! Doug On Mon, 26 Jun 2006, Iain Barker wrote: Doug, What you are describing sounds like the Aastra 480-CT, a base Ethernet/SIP screenphone supporting multiple wireless handsets [but as this is a non-commercial list I won't go into more detail here, google for the above model number if you're interested in more info.] - Iain --- Message: 4 Date: Mon, 26 Jun 2006 00:08:48 -0400 (EDT) From: Doug Crompton [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] best hardphone for Asterisk? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII Still awfully pricey for home use and the styling is not there for a bedroom or many other areas of a modern home. What we need is a wireless sip phone modeled like the panasonic or uniden which allow multiple extension off of one base. The base would connect to the internet. The other problem is many of these phones require power, so even if you have backup for your central system the phone still needs to be on it. Power over ethernet would help. Doug Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Question about ring groups and ext. busy in call
There is a difference between call waiting (a single line
with multiple call instances) and a multi-line phone.
We use a Polycom phone that has 3 lines with a single call
instance per line. The configuration is set so that once a line has a single
call it will return busy.
Therefore, we have use the local channels to acheive the
functionality that you are looking for as outlined below. It results in a number
of macro's being run in parallel, but as long as you have enough horse power it
shouldn't be a problem.
Hope that helps.
Cullin J.
Wible
Co-Founder
CTO
Email Data Source,
Inc.
212-514-8900
x1006
exten =
all,1,Dial(local/SIP-0004f2026a53local/SIP-0004f2035e5f)
exten =
SIP-0004f2026a53,1,Macro(exten-chain,SIP/0004f2026a53-1,SIP/0004f2026a53-2,SIP/0004f2026a53-3)
exten =
SIP-0004f2035e5f,1,Macro(exten-chain,SIP/0004f2035e5f-1,SIP/0004f2035e5f-2,SIP/0004f2035e5f-3)
;; Ring a chain of two devices (no
voicemail):; ${ARG1} - Device(s) to ring;
${ARG2} - Device(s) to ring (when busy); ${ARG3} - Device(s) to
ring (when busy);[macro-exten-chain]exten =
s,1,Playtones(ring)exten = s,2,Dial(${ARG1}, 30,
r)
; do the callexten =
s,3,Goto(207)
; error, busy
exten = s,103,GotoIf($[ "${ARG2}" != "" ]?104:207); try
the 2nd deviceexten = s,104,Dial(${ARG2}, 30,
r)
; do the callexten =
s,105,Playtones(busy)
; play busyexten =
s,106,Busy()
; mark busy
exten = s,205,GotoIf($[ "${ARG3}" != "" ]?206:207); try
the 3rd deviceexten = s,206,Dial(${ARG3}, 30,
jr)
; do the callexten =
s,207,Playtones(busy)
; play busyexten =
s,208,Busy()
; mark busy
exten = i,1,Playtones(busy)exten =
i,2,Busy()
exten = t,1,Playtones(busy)exten =
t,2,Busy()
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
SuttonSent: Monday, June 26, 2006 11:15 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Question
about ring groups and ext. busy in call
I have a ring group set up with 3
extensions well use 14, 15 and 16.
When a call comes in, it rings all
three extensions. If one particular extension already is on the phone, it
completely skips that phone and only rings the other 2. Example to
explanation sake is:
Call comes in, ext. 14 is already in
the middle of a call, 15 and 16 will ring normally, but 14 does not have any
indication that another call came in.
What Im trying to accomplish
is:
I would like the ring group to
always ring all 3 phones, even if one is on the phone. Similar to call
waiting I guess on the multi-line phones, it could ring line 2 or 3 or which
ever line is available if that phone is already in
progress.
This is necessary because some times
there is only 1 person in the office and may not always be able to hear the
other phones ringing
Call pick up is not what Im looking
for.. (mainly because again, the person may not hear/know the other phones are
ringing).
Thank you for any
help!
Chris
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RE: [Asterisk-Users] Looking for SIP provider with minimal call setuptime
Use Teliax - http://www.teliax.com/ Cullin J. Wible Co-Founder CTO Email Data Source, Inc. 212-514-8900 x1006 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Arnaud Sent: Monday, June 19, 2006 7:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Looking for SIP provider with minimal call setuptime I am looking for a PSTN-SIP provider with minimal call setup time. That is When a POTS phone calls my SIP number, RTP traffic between the SIP provider and my SIP phone should start flowing ASAP. SIP client uses auto-answer. I tried IPKall that resulted in intermittent total failure. With Broadvoice plan at $9.99 call setup time is about 7 seconds. We have a fast IP network here , so I am quite sure that the limiting factor is the SIP provider, not bandwidth or RTT. BTW RTT to broadvoice proxy was 20ms. US west coast number prefferred. If _you_ are a provider, please contact me off list. thanks - Arnaud ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A2Billing
Not that this is an ideal situation, but can you wrap the php code in a
shell script and trap the HUP signal?
Just a thought.
Cullin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Areski K
Sent: Wednesday, November 09, 2005 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] A2Billing
Well what is happening is that Asterisk is sending a SIGHUP signal to
the AGI script.
I guess this feature was always implemented in Asterisk but wasn't
working before
and has been corrected with the pthread_sigmask(SIG_UNBLOCK (see
res_agi.c)...
PHP CLI/CGI doesn't normally include the pcntl module for signal
handler so the SIGHUP
is killing the AGI process, snifff ;(
So what the solution for those that want to use the cvs or start to
enjoy 1.2.0-rc1, well
there 2 solutions :
1# Remove the kill(pid,SIGHUP) in res_agi :D Kidding, some would like
to kill me for saying smth like that.
2# install php-pcntl according to your php version (PHP 5 have it already)
- u can download from here
http://sourceforge.net/project/showfiles.php?group_id=112092
and then add this at the beginning of a2billing.php (after
#!/usr/bin/php -q ?php... )
declare(ticks = 1);
if (function_exists('pcntl_signal')) {
pcntl_signal(SIGHUP, SIG_IGN);
}
This will ignore the signal SIGHUP.
Anyway, I will release tonight or tomorrow (if I am lazy) so you can
catch this changes from the packages.
Don't bill too high save the killed process,
/Areski
http://areski.net/a2billing/
On 11/9/05, Administrator TOOTAI [EMAIL PROTECTED] wrote:
John Fraser a écrit :
Hi all,
I am having an issue with individual access vs simultaneous access.
If I set a card for individual access, make a call with that card the
counter
goes to 1. If the call complets normally shouldnt the counter reset to
0?
Second call tells me that card is already in use.
If you're using CVS/1.2, A2Billing is broken and don't recognize hangup.
It's ok with 1.0 branch. Other solution is not to hangup and let
A2Billing do the stuff ;-)
[...]
--
Daniel
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RE: [Asterisk-Users] sill looking for a provider
We have been using Teliax (www.teliax.com) for a while now and have 3 accounts with them (one for each of our asterisk servers). They've had their ups and downs but have been working to improve their support and now we are now able to speak with someone during their business hours (8-5PM MST). There was recently an issue with a high-latency link between our backbone and their backbone providers. We called them about it to find out that they had already escalated the issue with the backbone carrier and the issue was resolved. The long and short of it is that when we have had issues (which are few and far between) they have been quickly resolved. There is no question in my mind that a growing industry and growing companies are going to face customer service issues. It's bad for us now, but I think it's a good sign for the open source VoIP community. I think patience is key. Originally we looked at a number of carriers, based on the following requirements: 1) Be located in the US. 2) Have a customer support phone number and answer the phone. 3) Accept major credit cards and automatically bill my account (no need to recharge via pay-pal). 4) Allow for business/corporate usage. 5) Support IAX and g726/g729 codec's 6) Support Set Caller ID 7) Support multiple-inbound DID's We didn't care about call forwarding, voicemail, 3-way calling or any of the other features that must residential carriers tout as features. I wish I had done my research a but more formally, but the answer after about a week of research and test accounts was to use Teliax. Hope that helps. Cullin J. Wible President CEO Algorim Technologies, LLC 212-535-3238 x102 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Piotr A. Sygula Sent: Friday, November 04, 2005 6:44 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] sill looking for a provider That concept is not bad; except when the CEO from the same company as the tech that calls all the time happens to call you from what appears to be the same caller id, and the CEO ends up hearing rap or hard rock... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Sent: Friday, November 04, 2005 5:32 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sill looking for a provider Jason Brashear wrote: Is there a provider that has good support and answers the phone? (= I need to get lines for my Asterisk server and want to move from broadvoice.com. So far I haven't been able to get anyone on the phone. Too funny... I was able to get them on the phone today but it means waiting on hold a very very long time. Maybe I should look for a provider that uses good quality comedy instead of music on hold? Even better we could add a feature to asterisk where you set your preference. Press 1 for rock, 2 for rap, etc. and the system uses your caller ID to remember that for subsequent calls. The latest acronym is the industry is HOIP. That stands for hold over IP. Rumors are that it will be patented in the US soon. You've been a great audience. Thank you very much. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: [Asterisk-biz] Asterisk as a Voice ConferenceServer
We used to run a conference server on a PII 400Mhz with 512MB of RAM. We had 2 separate conference rooms with 15 users each (30 simultaneous) calls with no problem. We have since upgraded it to a P4 2.0Ghz with 1GB of RAM (just because it was getting old) and it still works just fine with even higher call volumes. No degradation of quality either that we can see. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Wednesday, November 02, 2005 2:51 PM To: Iain Barker Cc: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] RE: [Asterisk-biz] Asterisk as a Voice ConferenceServer Iain Barker Wrote: - Our experience with over 10 or more participants in a single Asterisk conference was that quality degraded quite rapidly. Is this really true as there were many in this list who had confirmed that they have used the conference bridge for a lot more connections than what you have Suggested as the upper limit. Logically the conference bridge should work at the same capacity as the number of calls Asterisk can handle in a given configuration. Though your solution looks impressive and probably is the best for upto 30 simultaneous calls, I am more interested in knowing what it takes for Asterisk to be able to handle the 100 channels I need to run Simultaneously. Seshu Kanuri -Original Message- From: Iain Barker [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 02, 2005 1:41 PM To: Kanuri, Seshu (Company IT) Subject: Re: [Asterisk-biz] Asterisk as a Voice Conference Server Seshu, Our experience with over 10 or more participants in a single Asterisk conference was that quality degraded quite rapidly. The solution was a dedicated hardware bridge for conference mixing http://www.aastra.com/enterpriseip/pro_238.asp Kanuri, Seshu (Company IT) wrote: I am working on a bid for a New York State requirement where we need to provide access to 100 Simulataneous Investors to get into a conference with the Pension Funds Officer for discussions. As you might have guessed it, I am presenting an Asterisk enabled Conference solution. One of the Bid requirement is to provide three verifiable references who have implemented a similar voice conference solution for more or less 100 simultaneous calls, with a possible recording of the entire call. If anyone has implemented this on a commercial scale, I am looking for referrals at this time, and a possible co-operation in future. I would appreciate if you can send me your name, contact Info, company name and a one para description of the solution and the name/type of client whom/where this solution is running at this time. A couple of minutes of your time is needed when the guys at Albany may like to speak to you for a confirmation that Asterisk is real and it can do the 100 people conference, what they are looking for. I do thousands of conferences a day using asterisk as the backend, most are in the 5-50 user range, but many are in the 150+ range. (but, I use app_conference, not app_meetme for them). I can give you my contact information off-list if you want it. -SteveK NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail -- newbie question
1) Create the following in your dialplan: exten = 100,1,VoiceMailMain() 2) Set their password to 1234. They can change it in the voicemail menu. 3) See: Getting MWI on Polycom Phones to work with Asterisk http://www.voip-info.org/wiki-Getting+MWI+on+Polycom+Phones+to+work+with+Ast erisk I don't know about Firefly. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rudolf Ladyzhenskii Sent: Saturday, August 06, 2005 7:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Voicemail -- newbie question Hi, all I am trying to set up voicemail. I've done it to the point where I can leave messages. How do I retrieve them? Actually I have few questions: 1. I want voice mail to be available at certain extension, say 100. How do I set it up so all users can call this number and get to their respective mailboxes. 2. How do I let users to create their own voicemail passwords from the phone? 3. How do I tell users that they have message? I use Polycom SP300 phones and FireFly IAX phones. I can do it via e-mails, but prefer visual indication on the phone. I have looked at wiki, but did not find answers to all questions. Is there a voicemail setup for dummies type of resource? Any help is appreciated. Thanks, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Phones
A few comments: 1) We are using quite a few SoundPoint 300 and 501's with no problem. 2) Your intermittent ring issues sounds like what we saw when we tested the phones through NAT (which doesn't really work at all despite what the documentation says). 3) We also upgraded to the latest boot rom and firmware before testing. See: http://www.freedomphones.net/polycom/files/ We also set the call count for each line to 1 so that once it has a call going it is marked as busy causing asterisk to use the second call instance. We have a macro that tries all of the call instances for a given phone in order (the devices are marked MAC-1, MAC-2, etc). That way calls only ring line 1, then line 2, then line 3, etc. Try that and then see what happens. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Hayden Sent: Friday, August 05, 2005 7:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom Phones Uhmm. Well, he may just be using the typical asterisk configuration of just editing the .conf files rather than using AMP or [EMAIL PROTECTED] For the first issue, that kind of sounds like a problem with the polycom configuration. I don't have my pdf of the polycom config in it with me right now, but I'll bet there is a setting you can toggle to fix that on the boot server files (or maybe even on the phones config). What version are you using? Second, it sounds like you may be having problems with SIP or NAT or registration somehow. I'm curious to see the CLI output on that before I could diagnose it. Hope you like the Polycoms! I love them! I suggest you check out the wiki page on them - it's EXTREMELY helpful at getting things setup correctly with all the great features they built into those phones. -- Tom Polycom Fanboy Hayden Astoria Telecom, LLC www.astoriatelecom.net On 8/5/05, Ariel Batista [EMAIL PROTECTED] wrote: Chris Gamble wrote: Just got in a bunch of polycom phones for use on my shiny new asterisk box, but found 2 small issues I was wandering if someone could help me with. Are you using AMP or Asterisk @ Home? First, though the phones support 2 call appearances, if I am on a call, the second call does not ring through -- it goes to voicemail instead of letting me put the first on hold to talk to the second. Is there a way to fix this? If you are then you need to turn call waiting on * 70 The second is: a lot of my phones will not ring for internal extensions. They show up on the screen as a call ringing in, but the phone itself wont ring. About 50% however do ring. What could cause this? Are the phone registered correctly? What are the settings you have on them. As usual, thank you all for your kind support in getting this far! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call transfer
You must use the 't' 'T' options in the Dial() command when placing calls to and from the device. We had extensions that were combinations of SIP and IAX devices and didn't want/need this behavior on all of our devices so we setup our extensions with something as follows: Exten = 1000,1,Dial(Local/IAX-1000/[EMAIL PROTECTED]Local/SIP-1000/[EMAIL PROTECTED], 60, r) [devices] Exten = SIP-1000,1,Dial(SIP-XYZ, 60, tr) Exten = IAX-1000,1,Dial(IAX-ABC, 60, r) That will ring both devices using different dial statements for each. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, August 01, 2005 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] call transfer Hi! I have searched answer how can I transfer calls with asterisk,with no result. Can you advice me and show some example file how can I use SIP phone to transfer calls by hitting # and get the Transfer prompt and enter an extension I want to transfer to? Thanks for your answers This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail envelope time is 4 hours ahead
I had the same problem in 1.0.9. We fixed it by moving the [zonemessages] section above the [general] section so that it gets processed first. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Frank Tarczynski Sent: Monday, August 01, 2005 6:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail envelope time is 4 hours ahead I'm running a recent CVS build under Solaris 10. In the shell than I'm running the Asterisk console I have TZ=US/Eastern and in my voicemail.conf I have tz=eastern and eastern=America/New_York|'vm-received' Q 'digits/at' IMp. The voicemail envelope information seems to be exactly 4 hours ahead. No matter what I try I can't seem to find the cause. Any ideas? Frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom digitmap question
It is my understanding that the purpose digitmap is to determine when the phone should transmit the digits entered to the server. I do not believe that it has any method for changing the dialstring. However, you could place the Polycom phones in their own context which would perform this mangling for you. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Tuesday, July 26, 2005 4:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom digitmap question [EMAIL PROTECTED] wrote: via google, I found the reference regarding digit maps for the Polycom phones: http://lists.digium.com/pipermail/asterisk-users/2005-January/082884.html But I don't see any instruction for prepending digits to the number dialed. Does anyone know how to prepend a digit to the number dialed (from the Polycom side, not Asterisk)? I can do this pretty easily on a Sipura. i.e. Say I want to add the digit 9 to what the user dials 1xx, the Polycom should actually send 91xx to Asterisk. I've never seen this option in the Polycom Admin Guide (which doesn't say much about the digitmap and refers you to the MGCP RFC digitmap handling. --Eric -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Generate ring while waiting for SIP connection toinitiate
Use the r option in the Dial() command. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Kartsioukas Sent: Tuesday, July 26, 2005 4:57 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Generate ring while waiting for SIP connection toinitiate We're passing PSTN traffic on to a SIP proxy. The SIP phone customers have voicemail that will answer if their phone isn't picked up in a certain amount of time. However, if their phone is not on the network, a caller will get nothing but dead air as Asterisk keeps attempting to initiate the SIP connection. Is there a way to generate a ringtone for the caller while Asterisk is trying to make the SIP connection? -- Nick Kartsioukas Sky Way Networks, LLC ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Automatic setup of calls between two externallines
I think you could accomplish this with EAGI or the manager interface. You should also read: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out On a side note, we spent lots of time when we setup our Asterisk system dealing with the answer detection for PSTN calls. We use Teliax for VoIP traffic as well as a 4-port Digium FXO card. We have come to understand that POTS lines can not accurately handle answer detection and that some of the 800 numbers (e.g. American Airlines, Staples, etc.) also pose problems even for all-digital services (including T1 carriers). We have had the most success placing PSTN calls if we Answer() the call first and then Dial() the PSTN telephone number. This also allows VoIP users to accurately hear telco messages (such as this number is not in service) that happen before the channel is answered. After weeks of debugging and testing I think that accurate answer detection on the PSTN will be the most complicated part of what you are attempting to do. Granted if you are calling a limited set of numbers you can probably test it and see how it works but I would proceed with caution. And just remember that this has nothing to do with Asterisk it is an unfortunate part of the PSTN. The better solution would be to call user 1 and play a repeated prompt saying Press # to initiate your connection. Then on DTMF# you would initiate the second call. It's not elegant but it's bullet proof. Hope that helps, Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Billy Dunn Sent: Tuesday, July 26, 2005 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Automatic setup of calls between two externallines Rob Scott wrote: Is it possible to automatically set up a call between two external lines? I would like Asterisk is call a cellphone number, wait for it to answer, and then call another cellphone, when that answers connect the two together. I assume it is possible but can someone point me how to do it. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I think this is possible, but the conversation will look like this: 1st cell phone -- asterisk -- 2nd cell phone ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mpg123 - two processes
I always have two as well - not sure why though. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Billy Dunn Sent: Tuesday, July 26, 2005 6:25 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] mpg123 - two processes Does everyone have two processes running for mpg123? I always have them when I'm running an idle Asterisk box. No calls going in or out and nothing off hook. Is this normal? Thanks! 5008 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-ri 5015 ?S 0:00 /usr/sbin/asterisk 5061 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-ri ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Voicemail Send Message (Options 3, 5) Patch
From: Cullin J. Wible [mailto:[EMAIL PROTECTED] Sent: Monday, July 25, 2005 3:36 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'; 'asterisk-dev@lists.digium.com'Cc: 'David Rule'Subject: Voicemail Send Message (Options 3, 5) Patch We run Asterisk 1.0.9 with multiple voicemail contexts and realized that there was a bug in the Voicemail Send Message feature that allows you to send a message to another user. We have fixed the bug with a patch and thought that other people might want to see it. The problem is that the context was being left off of the destination extension when you entered a voicemail message to forward to. I hope this helps. Cullin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call forwarding
1) You could use asterisk realtime and a mysql database.
2) You could use an asterisk database and allow users to set call forwarding
by calling an extension.
3) You could write some scripts to use an external database (what we did)
and either allow users to update their forwarding options via a web page or
telephone.
I have attached some simple shell AGI-scripts and parts of our dial-plan so
you can see how it all works. We authenticate against the mysql voicemail
database and then our standard extension macro checks the database, possibly
adding another channel to the dial command.
I hope this helps.
Cullin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian A
Sent: Monday, July 25, 2005 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Call forwarding
Is there an easy way to allow the users to go to a webpage or dial an
extension and enter a phone number that their extension can be
forwarded to?
I'm using SER+Asterisk so doing this in sip.conf for example would not
work since all users are registered to SER. Currently in
extensions.conf I have:
exten = s,2,Dial(SIP/[EMAIL PROTECTED],20)
Is there a way to check that the user at ${ARG1} has setup forwarding
and retrieve the forwarding destination?
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extensions-vmauth.conf
Description: Binary data
forward-get.agi
Description: Binary data
forward-set.agi
Description: Binary data
voicemail-auth.agi
Description: Binary data
extensions.conf
Description: Binary data
extensions-forward.conf
Description: Binary data
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RE: [Asterisk-Users] Unable to call certain 800 numbers through Teliax
After digging into this further, it appears that removing the r from the Dial plan doesn't fix the problem. While removing the r option allows you to hear the answered message at the remote end Asterisk does not consider the call to be answered so the extension will move onto the next priority when the Dial timeout is reached. I have reproduced this problem now directly with a Zap channel and a FXO POTS device. So I think this is probably a but with the answer detection in the Zap channel device driver. Does anyone have any thoughts before I report this as a bug? Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cullin J. Wible Sent: Thursday, July 14, 2005 11:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: 'David Rule' Subject: RE: [Asterisk-Users] Unable to call certain 800 numbers through Teliax After all of your feedback and a discussion at Teliax we have fixed this issues. It appears that when dialing a PSTN number, using the 'r' option is really unnecessary. Furthermore, some IAX clients and older phones (e.g. Cisco 20 VIP) require us to Answer() the call before dialing the PSTN network or Teliax. For more information, see the thread on Teliax at http://www.teliax.com/forum/viewtopic.php?p=544#544. Thanks for all the help! Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, July 13, 2005 12:38 PM To: asterisk-users@lists.digium.com Cc: asterisk-users@lists.digium.com; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Unable to call certain 800 numbers through Teliax We are unable to call certain 800 numbers through Teliax but I thought I would post this here and see if anyone else had the same problem with either Teliax or other carriers. The 800 numbers causing problems pick-up the call right away and are in the US - American Airlines (8004337300) and Staples (800-378-2753) - we can call many other 800 numbers just fine. My users have reported the same problem with AA, we also use Teliax. I coul care less about Staples but American Airlines is the airline that serves this destination, so it is important to us. I'm not the OP, but I tested both the AA and Staples numbers again this morning via teliax. Still working just fine here (C7960, cvs-head from last night). So, if its not working for both of you, the problem must be: - already fixed in asterisk head, or, - the iax2 call termination equipment (not necessarily asterisk) used by teliax to complete your calls is different from my calls. We use teliax and I had a similar problem with UPS. I can currently call Staples and AA fine. The problem was with numbers that did not generate a ring tone before answering. I solved this problem by changing my Dial command for outbound. I had the 'r' option in there before, so essentially the number would just keep ringing to the user, while on the other end it had actually answered. If this is not your problem, please specify in more detail the behavior you are seeing. What is the output on the asterisk console when one of these calls is made? What version of Asterisk are you using? -Ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to call certain 800 numbers through Teliax
After all of your feedback and a discussion at Teliax we have fixed this issues. It appears that when dialing a PSTN number, using the 'r' option is really unnecessary. Furthermore, some IAX clients and older phones (e.g. Cisco 20 VIP) require us to Answer() the call before dialing the PSTN network or Teliax. For more information, see the thread on Teliax at http://www.teliax.com/forum/viewtopic.php?p=544#544. Thanks for all the help! Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, July 13, 2005 12:38 PM To: asterisk-users@lists.digium.com Cc: asterisk-users@lists.digium.com; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Unable to call certain 800 numbers through Teliax We are unable to call certain 800 numbers through Teliax but I thought I would post this here and see if anyone else had the same problem with either Teliax or other carriers. The 800 numbers causing problems pick-up the call right away and are in the US - American Airlines (8004337300) and Staples (800-378-2753) - we can call many other 800 numbers just fine. My users have reported the same problem with AA, we also use Teliax. I coul care less about Staples but American Airlines is the airline that serves this destination, so it is important to us. I'm not the OP, but I tested both the AA and Staples numbers again this morning via teliax. Still working just fine here (C7960, cvs-head from last night). So, if its not working for both of you, the problem must be: - already fixed in asterisk head, or, - the iax2 call termination equipment (not necessarily asterisk) used by teliax to complete your calls is different from my calls. We use teliax and I had a similar problem with UPS. I can currently call Staples and AA fine. The problem was with numbers that did not generate a ring tone before answering. I solved this problem by changing my Dial command for outbound. I had the 'r' option in there before, so essentially the number would just keep ringing to the user, while on the other end it had actually answered. If this is not your problem, please specify in more detail the behavior you are seeing. What is the output on the asterisk console when one of these calls is made? What version of Asterisk are you using? -Ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] extension mobility and CDR logging questions
While I have never done this, it appears that you could use the agents.conf to allow people to login to an extension and have calls forwarded to their current phone. The dialplan would then reference the Agent/XXX rather then the device they are working at. Hope that points you in the right direction. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of KRTorio Sent: Wednesday, July 13, 2005 7:39 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] extension mobility and CDR logging questions I intend to add to my asterisk system a feature similar to cisco call manager's extension mobility so that agents can log in to any phone in the office and keep their profile (ex. the agent's specific directory number). But before doing that, I need to confirm that asterisk doesn't have a native solution for that (ex. application/addon), and that nobody has come up with their own solution to it. Is it also possible for asterisk to include in the CDR any information about the agent who made/received the call? Ideas/suggestions are also welcome. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE: [Asterisk-Users] Any suggestions for an IP phone?
We debated using TFTP for our phones since that is what we already used for our Cisco 30 VIP phones. However, we decided to install FTP instead since the Polycom's have the ability to write their log files and config modifications back to the server, which is very helpful and we weren't going to enable write-based tftp access on our network. FTP also allows the phones to check the modification dates on the files to determine what has changed. Unless there is a good reason not to, I would use FTP. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, July 13, 2005 12:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: RE: [Asterisk-Users] Any suggestions for an IP phone? Polycom does not support Asterisk. Thsi does not mean phones do not work with it. Rudolf P.S. I am having troubles setting up Polycom 300 with tftp server. By some reason phones always say can not contact boot server. Phones are set to use tftp and correct boot server IP is set via dhcp. I will investigate further, but any suggestions are appreciated. List Receiver [EMAIL PROTECTED] wrote: According to voipsupply.com http://www.voipsupply.com/product_info.php?cPath=95_112products_id=817 --Please Note: Polycom phones are not supported under Asterisk Open Source PBX. Polycom certified platform partners include Path Navigator, Broadsoft, Interactive Intelligence, Sphere, Sylantro, Vertical Networks, VocalData, Alcatel and 3COM. For more information on Polycom supported IP Communications platforms-- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cullin J. Wible Sent: Tuesday, July 12, 2005 7:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Any suggestions for an IP phone? We just purchased 4 of the Polycom SoundPoint 301's. We are very happy with them so far. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, July 12, 2005 8:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Any suggestions for an IP phone? Polycom SP300 is a pretty good phone. Rudolf Alexandre Leclerc [EMAIL PROTECTED] wrote: Hi all, We are in the process of selection IP Phones to work with our *new* Asterisk PBX. We want to buy 4 for something less than 1000$ but with a nice set of features to work with our mail box, lines, good sound quality, full duplex (and maybe speaker phone). Any suggestions for something with good voice quality and not much troubles to setup with Asterisk? Voici quality is the most important point. Thanks for any sugestion. -- Alexandre Leclerc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to call certain 800 numbers through Teliax
We are unable to call certain 800 numbers through Teliax but I thought I would post this here and see if anyone else had the same problem with either Teliax or other carriers. The 800 numbers causing problems pick-up the call right away and are in the US - American Airlines (8004337300) and Staples (800-378-2753) - we can call many other 800 numbers just fine. Our asterisk setup has a 4-port digium analog card as well as a Teliax account, currently connected via IAX2. We are able to call the 800-numbers above through the digium card but not through Teliax. We have also tried calling the numbers with Asterisk and SIP through Teliax with no-luck either. We have also tried using several IAX soft-phones (using iaxComm) directly with Teliax account (no Asterisk) and have the same problem. The *only* way that we have been able to call these numbers through Teliax is to directly X-Lite directly connected to Teliax through SIP - so Teliax claims that it is not a problem with their service, but with Asterisk. In examining the asterisk log output it appears that the channel is never answered and continues to ring indefinately. My guess is that there is a timing issue with the signals being sent and recieved by Asterisk and Teliax and that the "answered" is probably being transmitted before the "ringing" signal and is therefore being ignored by asterisk and iaxComm. If anyone could give this a shot and see what works or doesn't work that would be great. We have reproduced this with Asterisk 1.0.7 and 1.0.9. Thanks, Cullin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Any suggestions for an IP phone?
We just purchased 4 of the Polycom SoundPoint 301's. We are very happy with them so far. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, July 12, 2005 8:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Any suggestions for an IP phone? Polycom SP300 is a pretty good phone. Rudolf Alexandre Leclerc [EMAIL PROTECTED] wrote: Hi all, We are in the process of selection IP Phones to work with our *new* Asterisk PBX. We want to buy 4 for something less than 1000$ but with a nice set of features to work with our mail box, lines, good sound quality, full duplex (and maybe speaker phone). Any suggestions for something with good voice quality and not much troubles to setup with Asterisk? Voici quality is the most important point. Thanks for any sugestion. -- Alexandre Leclerc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NO calling tone
Add the r parameter to the end of the Dial() statement. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Wong Sent: Tuesday, July 12, 2005 10:38 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] NO calling tone Hi, When I make a call by using sip phone or softphone, there is no calling sound, how do I get the calling sound ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
