RE: [asterisk-users] snom led not working with asterisk 1.4.1
Just to let you know that this doesn't work with the latest SNOM firmware. We use 6.2.3 which works fine. -- Jamie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carsten Bock Sent: 27 March 2007 12:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] snom led not working with asterisk 1.4.1 Steve Murphy wrote: On Wed, 2007-03-21 at 15:00 +0100, Giorgio Incantalupo wrote: Hi Steve, as you know if you type show hints inside asterisk console you can see phone status. When a phone is not connected, Asterisk says it is Unavailable. With Asterisk 1.2.9.1 my SNOM leds worked well so I knew when a phone was not available but with Asterisk 1.4.1 is not possible anymore. This is one of the functions which I'm trying to keep from Asterisk 1.2.9.1 to 1.4.1 . Pardon my ignorance! I am new in this area. I have not used my SNOM 360 with anything but 1.4. When the monitored extension is busy, the LED is on; when the extension is ringing, the LED flashes. What does it do for you in 1.2, when the line is unavailable? The LED is also on. I noticed a change from 1.2 to 1.4: channels/chan_sip.c, 1.2.13 : case AST_EXTENSION_UNAVAILABLE: statestring = confirmed; local_state = NOTIFY_CLOSED; pidfstate = away; pidfnote = Unavailable; Asterisk 1.4.2 channels/chan_sip.c Line 6892 function static int transmit_state_notify(...) case AST_EXTENSION_UNAVAILABLE: statestring = terminated; local_state = NOTIFY_CLOSED; pidfstate = away; pidfnote = Unavailable; The var statestring has changed. I changed it back to confirmed and the phone shows the unavailable state. ciao, Carsten ___ --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] RE: Snom 360 Multiple calls on hold help
Thanks again for this new beta release, I couldnt of asked for a quicker response time, my hat is truly off to Snom for actually caring about the customer! I'll 2nd that, we use mainly Snom's now and its mostly down to the fact they provide excellent customer service and support. And they also make very good handsets :) ___ --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] Audio goes one way during the call for a fewseconds. Is it RTP, NAT, dyndns, or what it is?
We had very similar problems to this which drove us insane for ages. Basically we use VoIP trunks (SIP) for all our inbound + outbound calls. Call quality was good however we would get random problems where people could not hear us or us hear them for about 5-10 seconds at a time. After weeks of trying to get to the bottom of the problem it appeared our VoIP trunk provider (sentiro/sip2go) had ropey connectivity at best. We have since changed provider and now experience no call problems whatsoever (after running extensive tests to the sip host such as mtr etc.) Jamie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: 05 November 2006 13:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Audio goes one way during the call for a fewseconds. Is it RTP, NAT, dyndns, or what it is? Sounds like a bad Internet connection messing with the IAX jitterbuffer. Try running ping plotter from your location to your host, and see if it goes 'red'/down. On 11/3/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Hi everybody, I finally want to get rid of 1-way audio problem. Please help me here. I have 3 scenarios. 1. Audio is always one way. Caller who dialed can't listen the called party but called party can listen him. In this scenatio Asterisk is on dynamic IP with dyndns FQDN. sip.conf has externip = abc.dyndns.org and localnet = xxx.xxx.xxx.xxx entry. Trunk and extensions are SIP. Where is the voice getting lost from the called party? NAT is there but Asterisk is in DMZ. 2. Conversation is going fine when all of a sudden you realize that other parth has started saying 'hello, hello' because they can't hear you. But you are hearing them loud and clear. Now you are on static IP with dyndns FQDN. externip and localnet settings in sip.conf (do we need them for static IP?). After about 15-20 seconds, again 2-way converstaion is established again. IAX trunk, SIP extension, no NAT. 3. Conversation goes one way for 15-20 sec during the most important part of the conversation (Murphy's Law). You are on a static IP with no dyndns enrty. Trunk is ZAP on PRI, extensions SIP. NAT present but router properly configures for port forwarding. externip and localnet settings present in sip.conf Is think may be due to some reason RTP stream gets lost, routed to wrong IP. But why would this happen during a call and how to stop it from happening. Or is there some other reason behind this? Does dyndns setting have to do anything with this problem? How can I overcome this problem once and forever. -- Zeeshan A Zakaria ___ --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] Example Polycom function key config
Jamie Heckford wrote: If it works during a call then excellent, I'll try have a play tomorrow and let you know how it goes as well. Have managed to get a dialstring sent now using speed dials, it also works incall! HOWEVER It doesn't seem to send them to the *current* call. It places the current call on hold and tries to place a call on a new line. Currently looking for a workaround to this, will let you know. Thanks, Jamie ___ --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: AW: [asterisk-users] Snom or Cisco Phones?
I really wouldn't bother with the polycom's if you want to be able to program the feature keys From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Schardin Sent: 03 November 2006 16:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: AW: [asterisk-users] Snom or Cisco Phones? Polycom with backite LCD (Is there any?) The Polycom 650 has a backlit display. They wont be shipping until some time this December. On Nov 2, 2006, at 7:51 PM, Zeeshan Zakaria wrote: Cisco is out of question because as somebody already said in this thread, they come with only half of the stuff, and then they are VERY propriatery. They'll give you really hard time in configuration, firmware upgrading, support etc. I'd say CISCO are not made for open source VoIP industry. My suggestion will be one of the Snom 360, Aastra 480i, Aastra 9133i or Linksys 942, or a Polycom with backite LCD (Is there any?) ___ --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 Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.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
RE: [asterisk-users] Example Polycom function key config
FYI - Polycom have confirmed to me that you can only send one digit via the programmable feature keys. Idiots. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jamie Heckford Sent: 01 November 2006 09:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Example Polycom function key config Hi Jamie - Hi Noah, Has anyone here reprogrammed their Polycom features keys using sip/ipmid.cfg? If so I would be really grateful if someone could send me an example Here's the keys line that I use for one of my clients: keys key.scrolling.timeout=1 key.IP_500.37.function.prim=DialpadPound key.IP_500.31.function.prim=DialpadStar key.IP_600.37.function.prim=DialpadPound key.IP_600.30.function.prim=DialpadStar/ Thanks for that, I have something similar but what I can't work out is how to send multiple digits. For example 2x 'DialpadPound'. I have tried putting it in twice etc. to no avail. Anyone know how to get this to work? I'm trying to get our transfer key (##) programmed to one of the function keys basically. Thanks, Jamie ___ --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] Example Polycom function key config
Jamie Heckford wrote: FYI - Polycom have confirmed to me that you can only send one digit via the programmable feature keys. Search the archives for the last month. If I recall correctly, if you are using firmware 2.0.1 it will allow you to map a speed dial and the speed dial can be programmed to send multiple digits. I'll be looking at this on Saturday. Doug Hi Doug, AFAIK (from looking through the archives) this will only allow you to send the digits onhook, not during a call. If it works during a call then excellent, I'll try have a play tomorrow and let you know how it goes as well. Jamie ___ --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] Example Polycom function key config
Hi Jamie - Hi Noah, Has anyone here reprogrammed their Polycom features keys using sip/ipmid.cfg? If so I would be really grateful if someone could send me an example Here's the keys line that I use for one of my clients: keys key.scrolling.timeout=1 key.IP_500.37.function.prim=DialpadPound key.IP_500.31.function.prim=DialpadStar key.IP_600.37.function.prim=DialpadPound key.IP_600.30.function.prim=DialpadStar/ Thanks for that, I have something similar but what I can't work out is how to send multiple digits. For example 2x 'DialpadPound'. I have tried putting it in twice etc. to no avail. Anyone know how to get this to work? I'm trying to get our transfer key (##) programmed to one of the function keys basically. Thanks, Jamie ___ --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] Example Polycom function key config
Hi, Has anyone here reprogrammed their Polycom features keys using sip/ipmid.cfg? If so I would be really grateful if someone could send me an example as I have tried various entries for hours now and don't seem to be getting anywhere. Any help appreciated. Kind regards Jamie Heckford Technical Consultant ___ --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] porting numbers in UK telewest/bt/adept
Any experts on porting numbers in the uk here? ;-) Yep, it is your legal _right_ to have the numbers ported in a reasonable time/cost. Point this out to them and ask what the complaints escalation procedure is. That should get their attention. Can you point me to the law that gives you the legal right to port numbers between providers? As far as I was aware they had to have a porting agreement with the new carrier to able to do this. ___ --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] Can anyone help? Why does One-Touch record mute/disconnect callif not dialed quick enough?
Hi, Any suggestions to below problem? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jamie Heckford Sent: 17 October 2006 21:48 To: asterisk-users@lists.digium.com Subject: [asterisk-users] FW: Why does One-Touch record mute/disconnect callif not dialed quick enough? Hi List, Have an odd problem with the one-touch record on asterisk 1.2.11. All works ok, however one of our users today discovered if he is a bit slow hitting the 1 key after he presses *, the call seems to stay connected but its almost like it is muted. Haven't figured out the delay yet but it seems to be if the 1 is not pressed within 1-2 secs this occurs. Any suggestions? I tried setting: disconnect = *0 in features.conf in the hope this would solve it but no luck. I am using Polycom SPIP 301 handsets and can't see anything obvious on these either/ Thanks in advance for any help! Kind regards Jamie Heckford Technical Consultant Interfuture Systems Ltd Kemps Farm Business Park, London Road, Balcombe, West Sussex RH17 6JH E-MAIL DISCLAIMER: This e-mail is intended for the addressee named above only, and may be covered by legal privilege and/or protected by law. If you are not the intended recipient please notify the sender immediately, and in the meantime do not disclose the contents to any other person nor use, copy or store the e-mail in any medium. As communications via the Internet are not secure Interfuture Systems Ltd can accept no liability if this e-mail is accessed by third parties during the course of transmission or is modified or amended in any way following despatch. Any views or opinions expressed within this e-mail are solely those of the sender, and do not necessarily represent those of Interfuture Systems Ltd unless otherwise specifically stated. Although Interfuture Systems Ltd has taken every reasonable precaution to ensure that any attachment to this e-mail has been checked for viruses, it is strongly recommended that you carry out your own virus check before opening any attachment, as we cannot accept liability for any damage sustained as a result of software virus infection. Interfuture Systems Ltd reserves the right and senders of messages shall be taken to consent to the monitoring and recording of e-mails addressed to members of the firm. ___ --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] FW: Why does One-Touch record mute/disconnect call if not dialed quick enough?
Hi List, Have an odd problem with the one-touch record on asterisk 1.2.11. All works ok, however one of our users today discovered if he is a bit slow hitting the 1 key after he presses *, the call seems to stay connected but its almost like it is muted. Haven't figured out the delay yet but it seems to be if the 1 is not pressed within 1-2 secs this occurs. Any suggestions? I tried setting: disconnect = *0 in features.conf in the hope this would solve it but no luck. I am using Polycom SPIP 301 handsets and can't see anything obvious on these either/ Thanks in advance for any help! Kind regards Jamie Heckford Technical Consultant Interfuture Systems Ltd Kemps Farm Business Park, London Road, Balcombe, West Sussex RH17 6JH E-MAIL DISCLAIMER: This e-mail is intended for the addressee named above only, and may be covered by legal privilege and/or protected by law. If you are not the intended recipient please notify the sender immediately, and in the meantime do not disclose the contents to any other person nor use, copy or store the e-mail in any medium. As communications via the Internet are not secure Interfuture Systems Ltd can accept no liability if this e-mail is accessed by third parties during the course of transmission or is modified or amended in any way following despatch. Any views or opinions expressed within this e-mail are solely those of the sender, and do not necessarily represent those of Interfuture Systems Ltd unless otherwise specifically stated. Although Interfuture Systems Ltd has taken every reasonable precaution to ensure that any attachment to this e-mail has been checked for viruses, it is strongly recommended that you carry out your own virus check before opening any attachment, as we cannot accept liability for any damage sustained as a result of software virus infection. Interfuture Systems Ltd reserves the right and senders of messages shall be taken to consent to the monitoring and recording of e-mails addressed to members of the firm. ___ --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] Polycom 301's drop last two digits of dialed number
Hi All, Having a rather annoying problem with the Polycom 301 phones, suspect it to be my dialplan. Basically if you lift the receiver off the handset and dial a number, it will not let you dial a number longer than 10 digits (Can see this being acceptable in US, but in UK its a right pain). As soon as the 10th digit is entered, it starts to dial and the number is invalid. If the phone is left on hook and the number is dialed, it works fine when pressing the 'send' key on the handset as it sends the whole number. Can anyone shed any light on this issue? I thought it could be asterisk is trying to Dial to soon so I added a Wait in the dialplan but it didn't seem to work. Kind regards Jamie Heckford Technical Consultant ___ --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
