similar to: dialing application - newbie question

Hi I am having alot of difficulty connecting to SIP providers (I am trying 3) and can't seem to find anything similar in the wiki or on the lists.....I can receive inbound calls fine however on placing an outbound call the calling phone never gets 'connected' but 2 way audio is passed for about 20secs before some sort of timeout. Anything suggestions as to what I could try

Hi Hope someone can help :) I am testing 4 PSTN termination providers. 3 SIP and 1 IAX IAX and 1 of the SIP providers work fine. Now the wierdness: 2 SIP providers I can only get oubound calls to ring at the destination and then nothing more. 1 gets as far as SIP code 183 (and ringing on the src handset ...yay) the other doesn't get past 100. Added to this inbound calls

Since 1.4 ALERT_INFO variable has been deprecated. I used to send this via AMI: Action: Originate Channel: Sip/1234 Application: AgentLogin Data: 1234 Variable: _ALERT_INFO=info=alert-autoanswer Callerid: AutoLogin[1234] In order to send an autologin and autoanswer call to the agent 1234 on an Aastra phone at extension 1234. (just for example). Now in * 1.4 with ALERT_INFO deprecated I

I have been playing with a 480i with the new firmware 1.2.0.162 I hope to get some form of paging intercom function to work. In the wiki someone post that ALERT_INFO type of paging might be in this version of firmware but I have been unable to find anything on this yet. I have tried sending the ALERT_INFO to the phone a number of ways with no results. I then hooked up my bt100 and tried to dial

But where do can you get this later firmware from? I'm still on 1.0.0.78 on my 480i. Regards Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Peter Passchier Sent: 05 August 2005 00:04 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] AASTRA 480i Firmware 1.2.0.162 SIP ALERT_INFOproblems

Hello List, I'd like to ask how much you guys pay for an E1 (30 voice lines) and what company. You can email me personally and not the list. Best regards, Eddie _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

Hi All. I am beginning a project of Call center and predictive diales, my call center have 50 operators, I have 50 analog phone line with the company PTT in my country. I have the following questions: 1- Can I to work this project with Asterisk? 2- What caracteristic of hardware need for my servers? 3- For 50 analog phone line what tipe of card digium I need? Thanks in advanced, Regards.

ok what softphone i should use to fit windows and linux supporting iax,thanks in advance. _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar – get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

often when I "try to" visit the scriptaculous, it crashes IE - does anybody else experience this as well? _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar – get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

If you have any quicknet cards you are not using, I may be interested in them. I'll discuss terms after I know what you have. For sake of high shipping costs, I'm not interested in overseas shipments to the United States (where I live). My best resources will be for those that have PCI PhoneJacks, or PCMCIA CardJacks. Please reply to my email address, and not the mailing list.

Hello, I'm working on a server and I enabled quota's on the filesystem. This works fine. But Samba doesn't see that quota. I have read that you need to compile samba with quota support. My problem is that I have a running Samba (from SuSe 10) and I don't know if there is quota support build in. Is there some way to see if quota support has been compiled in? Greetings, Peter

Hi all, I hope someone can help, as I have an urgent problem. I've got a production Asterisk server thats been deployed, but we are seeing a strange voice echo problem. There is about a 250ms echo for the users in the office, and they are hearing their own voice back at them. I'm running the CVS Head code, on RH9.0. This is on a P4 box with 2gb of memory. The client SIP phones are

Hi, I've just found: http://lists.digium.com/pipermail/asterisk-users/2003-June/014475.html which talks about ALERT_INFO and Cisco phones. How do I actually get this working and what does it do? Do I need to add anything to the configs for the phone or is it just a SetVar(ALERT_INFO=1) - which I tried and it seemed to do nothing at all.. Thanks Andy

Hello, I'm working on a quota system here at school. I want to execute a script everytime a user writes on the samba fileserver. This script checks some quota stuff. My question is, is it possible that a script is executed everytime a user writes a file to the samba server and if it's possible, how? Greetings, Peter Fortuin

I have an Asterisk box, 1.4.30 with a PRI. A Mitel 3300 is connected to the Asterisk box via SIP trunking. When a user calls from the Mitel through the Asterisk box the user can speak but can not hear the far end. But - when I route the Mitel user to echo() it works, send and receive. The Mitel user also can record and playback greetings. One thing I have noticed is that when the Mitel user

Greetings, I am trying to implement distinctive ringing on a Cisco 7960. I have tried setting alert_info to chirp1 or chirp2 before dialing the phone, but it has no affect. If you have successfully implemented distinctive ringing on a 7960, I would really appreciate seeing the snipit of code that works. Thanks in advance Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775)

Hi I am struggling with hardware choices to get started with. My options are narrowed down to SIP phones - Polycom IP500, IP600 and Cisco 7940G. of importance is: - functionality / integration with asterisk - headset functionality and use - voice quality - build quality Is there much of a difference between Polycom and Cisco? Scanning the group it looks like there may be slightly more

Hi all, Has anyone gotten custom ring tones to work using ALERT_INFO with the Cisco 7940 SIP phone? I've read the wiki, but just can't get this to work. I'm currently using the 7.2 SIP image. Thanks, Chris

Hi List, I have to compare a relationship between y and x for two locations. I found logistic regression fits both datasets well, but I am not sure how to test if relationships for both sites are significantly different. I searched the r site, however no answers exactly match the question. I used Tukey's HSD to compare two means, but the relationship in my study was not simply linear. So I

The subject says it all really. Question 1. Here is some code created to illustrate my problem, can anyone spot where I'm going wrong? Question 2. The reason I'm following a manual specification of models relates to the fact that in reality I am using mgcv::gam, and I'm not aware that dredge is able to separate individual smooth terms out of say s(a,b). Hence an additional request,