similar to: A call from a call file always does a redial?

Most people seem quite positive about Snom phones, I cannot share this opinion. The displays are dying quite often, and firmware is buggy. I have tried every firmware from 4.5 up to 5.x and 6.04 but keep having problems with phones locking up or rebooting during an ongoing conversation. REALLY annoying for a phone that is advertised / targeted as a business class phone

Hi, What is the best way to implement Automatic Redial on No Answer ? Looking at http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-12.txtI can see how Automatic Redial on Busy could (should) be done. How would you do it on No Answer ? Is there any event you should SUBSCRIBE to so that you're notified that you're callee is available ? What if you ask to be notified

Hi, Snom phones firmware 5.0 is now out. Try it if you like: http://www.snom.com/wiki/index.php/Main_Page. Regards, --------------------------------------------------------------------- Usman Tahir snom technology AG www.snom.com --------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list, To make outgoing calls by skype i would like to have our crm app create callfiles like we do for normal calls. If i read the instructions it says this : ---quote--- The syntax for making an outgoing call using Skype for Asterisk is as follows: Dial(Skype/[<originator>@]<destination>) ---unquote--- So i create a callfile that looks like this: --- Channel: SIP/228

Hi list! While looking for the Snom 190 I found another phone, the Elmeg IP 290 (www.elmeg.de). Looking at the pictures & the specs they seem to be very similar beasts but the firmware is supposedly not interchangeable. Does anyone know the difference between the 2, do they work with Asterisk? The weird thing is that Elmeg has similar phones with the Snom look but they are ISDN only (no

Hello, I'm trying to figure out how to change the redial, thus far if I hit redial it will redial the last called I made that was answered, not the last call I made that was not answer. I'm using Asterisk 1.8 Thanks, Motty

Has anyone ever tried to roll out a *66 auto-callback-redial feature on asterisk? I'm sure that implementing this for outbound Zap calls would be a nightmare, but what about something easier, like internal extensions? On my old Panasonic key system, it used to be such that, if the called extensions were busy, you could press 6 while hearing the busy signal, it would beep twice and hangup.

I was wondering if anyone knew of a way to create a busy-redial feature in the * dialplan? For example, you try to call 12125551212 but the number is busy, so you hang up and dial *XX12125551212 and hangup again, then * would continue to retry calling the number until either it rings or a timeout is reached, if it rings * then calls back the exten that made the *XX call and bridges the two

Hi All, I need a softphone (PC/Mobile) which does auto redial in any case (noanswer, answer, busy, congestion etc) after a given time interval. So if the time interval was 5 secs, it would dial last number dialled after every hangup (or every failure to dial). Does anyone know such feature in a softphone? -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi

Hi, We are experiencing this issue of redial dtmf tones generated randomly in the Voip calls, we have asterisk 1.4.21.2, dahdi 2.0.2.2 and we have dtmf as rfc 2833, we have a cisco router at the location Cisco 2431, 8FXS (only one FXS is used for Fax and rest are empy) connected to the netgear switch and all the phones are connected to this switch and there are no non sip devices in the

Hi, We have few systems with asterisk 1.4.22.1 and we use sip trunking for them not PRI's, one of our system is giving a problem of dtmf (rfc2833), like when we dial the number that have IVR and enter the extension or access code, it some time takes it and some times does'nt recognize the digits dialled. We also tried auto and info for dtmf but could not get the dtmf to work reliably, can

Hi list! Our CRM app is creating call files for outgoing calls which is working great I just have one problem. I am using this as my call file: Channel: SIP/228 (my phone) MaxRetries: 0 Context: from-internal (the context to dial from) Extension: 003120531234 (the phone number) Priority: 1 Callerid: Myfinecustomer <003120531234> so the external number is connected to my sip phone. However

Hi All, During last three month I have worked on improving functionality of Nortel phones working with asterisk to replace existing Nortel station by asterisk. Many improvments done, listed below. I have only i2002 phone and unable to test if new version of channel correctly works with i2204 phone. If anyone can test it and report issues, it would be great. Please visit mantis to find out patch

I'm rolling out an installation with snom 360s in the near future. Simple SOHO configuration, 3 FXOs hanging off a TDM400B, 4 snom 360s, a snom 200, some variant of IAX softphone, and an IAXy or Sipura 2002. I have the 360's set up to subscribe and notify for the line use lights, which works like a charm for interoffice calling (between the 360's, anyway. The IAXy, 200 and,

Would it be possible to use the programmable led+keys on the Snom phones to signal that there is an incoming call that is ringing a call group or pickup group? We use this on our existing PBX if for example the accounting dept. is out for lunch but nobody can hear their phones. This way you can see an incoming call (and we hate voicemail) :) Thanks!

I went to put in the new sound files over the weekend, but forgot to backup the custom folder and lost my custom digital receptionist files. I then had to copy the old files back from a duplicate machine. The problem is now though that voicemail just hangs up when I dial it. Other apps work - *70 for example gives me 'call waiting...activated' so I know it's accessing the files

Can I ask for some advice on dial-plan construction please I have setup my dialplan to use 9 to get a zap trunk, leaving everything else for internal extensions. However, this creates a problem in that my callerid is correct, but doesn't work to re-dial the incoming caller. So if I simply click "missed calls" on my Snom phone and hit redial then it tries to dial an internal

Hi list! I'm sure the topic has been discussed but I could not find what I was looking for. What would be the best / easiest VPN software solution. I would like to install vpn software on the * server for roadwarriors to connect to with laptops running windows. Ideally the vpn solution will not require any additional software on the client side but will use IPSEC. (Ofcourse call quality

Is it possible in any way to use an Outlook contacts database as the source for the internal Asterisk database that is used for callerid lookups? Thanks!

Hi I am trying to get a SNOM 360 to monitor other extensions i.e. when someone makes a call to/from another extension, one of the LED's on the SNOM 360 will change state. I am using 1.0.9/bristuff-8l. I have 2 extensions - 2001 is a SNOM 190, 2002 is a 360 - both are running the latest firmware. (3.60i for 190, 4.0 for 360). I have read all the relevant articles on the wiki on