[asterisk-users] seeding an originated number in a SIP phone [was: Re: Thunderbird extension using AMI to dial]
On Thu, Aug 25, 2011 at 07:36:53PM +0100, Chris Hastie wrote: > Hi > > I've just added direct support for AMI to a forthcoming version of > TBDialOut, a Thunderbird extension for dialling direct from > Thunderbird's address book. If anyone fancies testing it I'd be grateful > for any feedback. If you feel like casting a critical eye over the code, > or doing some translating, even better. > > AMI support is available in TBDialOut 1.7.0pre1, which can be found > either at http://www.oak-wood.co.uk/tbdialout/ or from the 'Development > channel' at the bottom of the page at > https://addons.mozilla.org/en-US/thunderbird/addon/tbdialout/ We already have a dialer script (sent to this list a while ago) so it's good to see that this extension support that simpler option as well (I don't use ThunderBird, as you can see. Some others in the office do use it). One followup question: I originate a call from a SIP phone to some remote number. The problem is that the number will not show up properly in the list of outgoing calls for the phone. Any idea how to fix this (for whatever SIP phone)? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_dahdi.conf waitfordialtone
has anybody made waitfordialtone in chan_dahdi.conf work outside the UK? callprogress=yes progzone=us waitfordialtone=1000 i keep getting: WARNING[3859]: chan_dahdi.c:7111 dahdi_read: Never saw dialtone on channel 1 i'm using asterisk 1.8.4.4 Kelvin Chua -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] USB or Ethernet based FXO device ?
I'm looking for an FXO device to connect to a POTS line that communicates via USB or Ethernet. Is there such a device ? Thanks ! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
Great discussion, people. I'm ordering hardware today. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
I gu On Thu, Aug 25, 2011 at 5:58 AM, Tzafrir Cohen wrote: > On Thu, Aug 25, 2011 at 12:39:14AM -0400, Steve Totaro wrote: > > > I used the Asteirsk System() app to call lynx with a special URL. The > URL > > contains all the authentication, recipient, and SMS body. Calling that > URL > > via System(), as I said, I like lynx, causes an SMS to be sent. Kannel > is > > extremely customizable. > > Slightly off-topic: why not use CURL()? > > -- > Tzafrir Cohen > icq#16849755 jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir > > Because my first inplementation was before Curl() was part of Asterisk. I am very familiar with Lynx and it wjust works. Never an issue. Don't enable or use apps on an Asterisk system if there is another, reliable app that can be used. I keep Asterisk's role to a minimum. I only load the apps that are needed for the implementation. I usually build them all, but, either I do a noload or rename the .so. I also try to put other functions of different machines, to segregate the mission critical, or at least the as much Asterisk from other features. Databases, fast-agi, HylaFax, recording calls, and whatever else. It is just the way I do things, budget providing of course. I want the core being as stripped down OS, apps, and Asterisk as possible. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to play wav files to all members in konference
Hi List, How to play wav files to all konference members at a time. I want to play with the help of AMI connection. I have tested that we can play channel base file playing. But it will take too much time if users are more then 20 - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound Dial
On Thu, Aug 25, 2011 at 5:18 AM, Kaushal Shriyan wrote: > On Wed, Aug 24, 2011 at 5:23 AM, Kaushal Shriyan > wrote: >> On Tue, Aug 23, 2011 at 12:09 PM, Faisal Hanif wrote: >>> U can also use VICIDIAL for it >> >> >> Hi Faisal >> >> Please help me understand the difference between VICIDIAL and >> astguiclient http://astguiclient.sourceforge.net. >> Are they both the same or interdependent on each other and also can i >> exclusively use it for my set up of 8 E1 PRI Lines meaning 240 bearer >> channels to make Outbound calls only and run a campaign --> meaning i >> have 200 phone numbers and a sound file of playtime of 30 secs. I need >> to dial out to all the 200 phone numbers and play the sound file >> concurrently at the same time. I have asterisk server version 1.8.5.0 >> on my server. >> >> Regards >> >> Kaushal >> > > Hi, > > Can someone please comment on my earlier email thread > > Regards > > Kaushal > Hi Again, Can someone please comment on my earlier email thread Regards Kaushal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?
On Fri, 2011-08-26 at 13:55 -0500, Ian Pilcher wrote: > > I don't have any experience with them, but the Siemens Gigaset A580 IP > seems to be about the best price point: > > http://www.voipsupply.com/siemens-gigaset-a580-ip I bought a Siemens Gigaset C475IP at the beginning of this year. RFC2833 dialling is broken, because it has broken handling of RTP mappings. If the peer doesn't ask for RFC2833 data in the same payload type as the C475 does (which it SHOULD but some don't), the C475 ignores the peer's rtpmap and just sends DTMF with the same payload type it asked to *receive* it in. I spent *months* trying to get a coherent response from Gigaset support and failing. Eventually when I filed a complaint about the lack of response, they told me they wouldn't fix it. They did say they'd replace it with a newer device but then have so far failed to follow through on that either. I won't be buying another Gigaset, and wouldn't recommend them either. -- dwmw2 smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?
On Sat, 27 Aug 2011, Alan Lord (News) wrote: On 26/08/11 19:02, linux guy wrote: I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home asterisk system. We've been using the Siemens Gigaset 685IP range for over three years and I'm (still) very pleased with them: +1 The base station is separate from the handsets - which is typically different from most DECT setups - the plus point is that you can position the base in a good location - ie. high on a wall, rather than anywhere else. Another plus is that the base has a single built-in ATA, so it can connect to the home PSTN line. The base also has an Ethernet socket to connect to the LAN and it can have up to 6 SIP accounts - each handset (up to 6) can be configured to ring on a particular SIP account or many SIP accounts and/or the PSTN line. Each handset has a default SIP account (or PSTN) to make outgoing calls on, but you can select any other SIP account or the PSTN by appending a code to the number you dial. They are very flexible - and being DECT, have superb range. I've installed many of these for my customers - typically the home office types - where they only want one phone on their desk - so the same handset can answer their home phone or their office SIP account, while providing wireless handsets throughout the rest of the house. A limitation is that one base can only handle 2 simultaneous SIP calls (plus a call via the PSTN), so if 2 phones are in-use, then the system can't take a 3rd call, however that's rarely a limitation in a domestic environment. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Integration with Android device
Hi, Have u tried NAT(Network Address Translation Settings) ?. place the following setting on SIP account. nat=yes externip=nnn.nnn.nnn.nnn ; externhost= localnet=192.168.1.0/255.255.255.0 externrefresh=10 for more details hv look on the http://www.wirelessforums.org/uk-telecom-voip/problem-getting-asterisk-behind-nat-run-sipproxd-56038.html Hope it helps !. Thanks, Ashik Ali On Thu, Aug 25, 2011 at 8:19 PM, amit anand wrote: > Hi > find the answer inline > > On Thu, Aug 25, 2011 at 21:24, Gopal krishnan > wrote: >> >> Thanks for all your comments. Actually I have 3G connection but even then >> the signal in my mobile automatically changes from 3g to 2G; it is >> automatically going to Edge signal. Anyways let me try with some other >> softphone like media5. >> Regards, >> Gopal >> >> On Thu, Aug 25, 2011 at 9:21 AM, James Perkins wrote: >>> >>> Try media5 fone. >>> I couldn't get 3cx to work on my iphone and tried about 7 different >>> softfones. Media5 is the best by a long shot. >>> Android is still in better and haven't tried it but if its anything like >>> their iphone app it will be worth a look. >>> There is a signup for the better at the website. >>> let us know how you go. >>> James >>> >>> - Original Message - >>> From: bakko >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Sent: Wednesday, August 24, 2011 11:48 PM >>> Subject: Re: [asterisk-users] Asterisk Integration with Android device >>> I think don't work with 2G network. >>> >>> Regards >>> >>> - Original Message - >>> From: Gopal krishnan >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Sent: Wednesday, August 24, 2011 4:01 PM >>> Subject: [asterisk-users] Asterisk Integration with Android device >>> Hi, >>> I created a extension in Asterisk, the extension has been configured in >>> Android softphone 3cx. When I tried to call from Andorid phone to some other >>> IP extension which is registered in Asterisk, I am not able to hear the >>> voice, when I check the asterisk log or wireshark there is only one way RTP >>> traffic, from Android I am connecting to Asterisk via 2G GSM network. >>> Any idea would be appreciated. >>> Regards, >>> Gopal >>> >>> >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > Try sipdroid for android > -- > > Amit Anand > > +91 9818559898 > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Trace to troubleshoot one way of communications
On Aug 26, 2011 4:54 PM, "bilal ghayyad" wrote: > > Hi All; > > How can I get a SIP trace to troubleshoot a one way of communications? I need to see what is happenning in the packets to know the reason of the problem. > Install ngrep on the box. Then type something like. ngrep -tq -W byline port 5060 Replace with the IP of the UA you want to monitor. Regards Jon Sent from my iPad3. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?
On 26/08/11 19:02, linux guy wrote: I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home asterisk system. We've been using the Siemens Gigaset 685IP range for over three years and I'm (still) very pleased with them: http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system
On 26/08/11 12:28, linux guy wrote: Great discussion, all of it. Thanks, people. How much power does the home asterisk box need ? Not much :-) I've been running our phone system and home media/storage network on a VIA C7 cpu based home build that I *downclocked* to 1Ghz from 1.2Ghz for about three years now. Al -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
