Re: [asterisk-users] problems with some incoming/outgoing calls
Hi, which version of asterisk are you running? Perhaps if you post your extensions.conf and others related files you could get more accurate help. If you answer a ringing phone and you can't answer the call, there you could have a network or sip config problem, that means that the SIP packet is not returning to the pbx. Regards. Lucas On Thu, 23 Oct 2008 10:30:29 -0200, Fernando Serto [EMAIL PROTECTED] wrote: Hi, I've been very puzzled lately. I installed a phone system for a friend a few weeks ago, and they're having a problem that I can't get rid of, actually 2 problems. Before I go into the problems, let me tell you about the setup. It's a pretty small setup with only 4 handsets, all Polycom 320s, the server is a Dell SC440 with Intel E2180 CPU (dual core, 2GHz) and 512MB Ram. Internet Connection is an ADSL2, with a not so reliable ISP in australia. For incoming calls, I had a Digium TDM410P with 4xFXO modules and HWEC. Because of these problems, i replaced the Digium card with a Sangoma A200D, but it didn't make any difference to the problems. All phones are hooked up to a Netgear PoE switch. Almost forgot to mention that this is not my first Asterisk setup, and in fact it is my 4th, and I used various SIP handsets before, and also different cards (Analog and Digital), so I'm not a total noob. Let's get to the problems... 1) Some incoming calls cannot be picked up Sometimes, incoming calls, coming through the analog card, cannot be picked up. All handsets are set to ring at the same time on incoming calls. and most of the time, calls can be answered on any of the handsets, but maybe 3 or 4 times a day, all handsets will be ringing, and you go to one handset to answer the call, you pick the handset, and it doesn't answer the call, it keeps ringing, then you go to another handset, and still can't pick up, sometimes, you can even try all 4 handsets, and no luck. but, at other times, you can't answer on the first handset, but you can on another, and it is totally random. but people are pretty pissed off for running around to answer a call. and what puzzles me is that you can sit around watching logs for hours, and it won't happen, other times, it happens 3 times in a row. any ideas? 2) Delay on outgoing calls via SIP People have been saying that when they call people, there's a delay for the call to be answered. For example, caller dials a number, callee answers the ringing phone, but caller is still listening to a ringing tone, and after a few seconds (up to 15 seconds) it sounds like the callee has just answered the call, when in fact, he had already answered a few seconds before. Problem with this is that some callees will hangup before the caller starts talking. These calls are going via pennytel, in australia, which seems to be a pretty good VOIP provider around here, and I've been using it on other setups and never had these issues. Well, sorry for the long first post, but I would really appreciate any suggestions you have. Cheers, Fernando ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cli with COLORS
Hi Everybody, I'm having a little problem with asterisk CLI, after the version 1.4.19 I'm not been able to see the CLI with colors anymore. I have a ubuntu box with asterisk 1.4.21 installed and I don't know how to enable the colors again. Of course I have the variable $TERM set to xterm-color. I would appreciate any help. Reagrds. Lucas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hook Flash
Hi, I'm having a problem conecting my asterisk 1.4.21 with zaptel 1.4.11 to a Panasonic PBX. I'm using dynamic features to send hook flash to the zap channels to make a call transfer to the pbx without tying a channel. When I call from asterisk to the Panasonic PBX I haven't any no problem, but when the call is from the Panasonic PBX, the dynamic features doesn't work. I have already tried all possible combinations in feature.conf: zapflash = *3,peer/both,flash zapflash2 = *4,callee,flash zapflash2 = *5,caller,flash In all cases I am setting the variable DYNAMIC_FEATURES before the Dial(). And is not a dtmf problem because I can see in the console the debug of the DTMF: chan_zap.c:1233 zt_digit_begin: Started VLDTMF digit '*' chan_zap.c:1268 zt_digit_end: Ending VLDTMF digit '*' chan_zap.c:1233 zt_digit_begin: Started VLDTMF digit '3' chan_zap.c:1268 zt_digit_end: Ending VLDTMF digit '3' The problem is that the application mapped in feature.conf it isn't been triggered. I would appreciate any help, I have already googled to death and I couldn't find anything. Thanks in advance. Lucas Alvarez -- ___ -- 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
Re: [asterisk-users] Hint extension issue - bug?
I'm using asterisk 1.2.10 David Gagnon wrote: Are you having this problem with the trunk? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Lucas Alvarez Envoyé : 22 août 2006 18:23 À : Asterisk Developers Mailing List; Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Hint extension issue - bug? Hi, I'm using the hint extension to monitoring the status of some extensions. If the extension is defined as a friend, the monitoring doesn't work any more. It only work if I define it as a peer. Is that right ? I mean, I supposed that an extension defined as a friend should have all the functionality of user and peer types. Is this documented somewhere? How can I know the status of an extension of type friend? I hope someone could bring me some light about this issue. Thanks in advance. Lucas Alvarez ___ --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
[asterisk-users] Hint extension issue - bug?
Hi, I'm using the hint extension to monitoring the status of some extensions. If the extension is defined as a friend, the monitoring doesn't work any more. It only work if I define it as a peer. Is that right ? I mean, I supposed that an extension defined as a friend should have all the functionality of user and peer types. Is this documented somewhere? How can I know the status of an extension of type friend? I hope someone could bring me some light about this issue. Thanks in advance. Lucas Alvarez ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Buddies in 1.6.6
Does someone know where I can get the last sip version? My Polycom reseller doesn't have it and I need to enable the buddy for 14 contacts. Thanks in advance. Lucas Alvarez Douglas Garstang wrote: I've never seen that problem, and I've only ever used 1.2+ with Polycom and buddies. -Original Message- *From:* Ryan Stark [mailto:[EMAIL PROTECTED] *Sent:* Tuesday, June 27, 2006 12:31 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] Polycom Buddies in 1.6.6 So I've got a 601 (1.6.6) with the side car, and the buddy watch seems to be working but it updates the statuses unreliably. When I do a sip show subscriptions in asterisk it lists my phone 12 times and at the bottom it says 0 active SIP subscriptions(s) I've got an older CVS-HEAD build, pre 1.2, do you think my problems are polycom or asterisk based? -Ryan On 6/19/06, *Kevin P. Fleming* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: - Douglas Garstang [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Polycom released their SIP software version 1.6.6 for their phones recently. I was under the impression that this release fixed a previous limitation where the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk. I have configured a phone directory to watch 30 or so appearances, and it still seems to only be sending 7 subscriptions to Asterisk. Has anyone else got this to work? Yes, it works on the Polycom 601 on my desk. However, the release notes say that the restriction was only removed for the IP600 and IP601; if you are using an IP300/1, IP500/1 or IP430 than the 7 buddy limit will still be in effect. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com http://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 -- NAT -- Internet -- NAT -- Sipura-3K (No Asterisk)
Joseph: I think that was under the sip tab you have to configure the sip proxy, just put the address of the firewall where you are doing port forwarding to the asterisk box. Also in the nat field at the sipura device put yes. And finally you have to open and forward the UDP ports of the firewall in the sipura LAN and forward it to the sipura device. Let me know if this works. Lucas Alvarez Joseph wrote: So far I have gathered that on my NAT (where asterisk server is) I have to port forward UDP ports: 5060 and range 1 - 2 to my asterisk server But I'm still stuck how to configure Sipura (behind NAT) what sip proxy and out-bound sip proxy, under which tab to change. ___ --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] UpState NY SIP provider
I'm using www.widevoip.com with asterisk, it works great. They provide us a VPN connection and a SIP user. Regards. Lucas Alvarez Andre Courchesne - Consultant wrote: Hi, Anyone has good/bad experience with SIP providers in upstate NY? Any recommendations of such provider who works great with Asterisk? Thanks, Andre Courchesne ___ --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] Registering Remote Sipura to Asterisk (both behind firewall)
Hi, you have to forward port 5060 udp and all the others udp ports that asterisk uses for RTP, 1 to 2 by default, see rtp.conf. At the sipura device in the server configuration you have to put the LAN ip of your Asterisk box as sip proxy server and firewall's ip as out-bound sip proxy. Joseph wrote: Can anybody point or provide working configuration how to register Sipura to Asterisk over the Internet. Both Sipura and Asterisk are behind firewalls. I'll be force to use SIP as that is the only protocol that Sipura is using. Do I need to enter any STUN Server: setting in SIP tab. On Asterisk I think I only need to make changes in sip.conf isn't it? What ports do I need to open on my firewall? ___ --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
