similar to: Consult transfer on SNOM 105

Hello all, I am running into some problems with a snom 105 phone trying to do a attended transfer . Snom phones are connected to Asterisk. This does not work, it will only do a unattended transfer. I have downloaded the manual from snom and followed the instructions. Has anyone experienced the same problem ? any ideas how to solve the problem. thanks, Arne.

Am 05.03.2015 um 15:09 schrieb James B. Byrne: > > On Thu, March 5, 2015 05:30, Ruben R?gels wrote: >> >> >> Am 05.03.2015 um 01:09 schrieb James B. Byrne: >>> I am trying to determine how the transfer button on the Snom-870 >>> works >>> with Asterisk. Is the ## special code employed as when it is >>> entered >>> through the

Hello, Before I dig into the SIP traces maybe someone has a clue: We have a few Snom phones, and the call-limit on the extension associated with them is set to 1 (so the other Snom phones can blink the relevant LED). When I call a SNOM phone (300 with the latest firmware) and the call is answered the phone is in "in use" state (verbose printing from Asterisk). When I hit the

Hello all! I have a few problems with Snom 320 phones: Problem A - Transfer out of Queues: We have a call center with some Snoms. We are using Queue and AgentCallbackLogin. As we run * 1.2.7.1 an agent could transfer a call out of the queue using the hold and transfer buttons on the Snom. This might have been the wrong way to do it all the time I found out later, but it worked. Now we upgraded

I am having a hell of a time with transfers. First the Snom issues: The transfer button on the Snom 220 does not work. I have read about setting break key off in the advanced page of the web config but the Snom 220 has no such option. At the moment I am having to use the # transfer hack which makes this phone look really stupid to have buttons on it that cannot be used. Anyone know how to

Hi, I found some issues with Snom 320 message retrieve key. This button works only when the MWI does not blink! If MWI blinks and I do press retrieve button I get "Unknown" on display and busy tone. From the sip debug it looks like that Snom does not send credentials to Asterisk which responds with 407 Proxy Auth required. I have loaded Snom with latest 5 firmware. No change. I'm

I would like to use one of my Snom 200's 5 programmable keys to park calls. I am using image SIP 2.04g. I have tried a variety of combinations and have come to the conclusion that: 1) On the Key Mappings administration page, I must select the "Transfer" under the "Break Keys" option box to be able to successfully transfer calls using the "Transfer" button. 2)

I wanted to post this to everyone in case someone else ran into this problem! Firmware 6.5.1 (Current stable release) Asterisk 1.2.13 Call Completion: Off Peer to Perr call completion: On (canreinvite=no is set in sip.conf for each friend) Call join on Xfer (2 calls): Off Those settings seem to do the trick, I tested on Snom 320s with no errors at all, Im going to test on the 360s later today,

Hi, I'm trying to set up my SNOM 200 with extensions, with different ringtones - but it doesn't seem to work. I've defined two extensions for it in Asterisk and in the SNOM 200 configuration. In the SNOM home>settings>SIP>Lines config page, I have set the ringer for the first extension to "ringer1", and "ringer6" for the second. In

On Thu, March 5, 2015 05:30, Ruben R?gels wrote: > > > Am 05.03.2015 um 01:09 schrieb James B. Byrne: >> I am trying to determine how the transfer button on the Snom-870 >> works >> with Asterisk. Is the ## special code employed as when it is >> entered >> through the handset or is the blind transfer through the phone >> function accomplished in a

I have setup an asterisk based phone system using snom-320 (SIP based) phones. I would like to change what seems to be the default procedure for an attended call transfer. Right now, the phone user places the call on hold, calls the extension using a "extension" button on the phone, speaks with the call recipient , and presses transfer to transfer the held call. The users would like

I am using the snom 200 with Phone type snom200-SIP Version snom200-SIP 2.04g Bootloader URL http://www.snom.com/download/snom200-boot1.9.bin Firmware URL http://www.snom.com/download/share/snom200-2.04o-SIP.bin I am using asterisk stable tree. I had to disable "Challenge Response on Phone" on my snom; I could not get it to work with

On Thu, March 5, 2015 09:56, Ruben R?gels wrote: > > Hi again, > > I'm glad to hear that I provided a somehow useful answer. > > Unfortunatelly, I don't know these details. > If you wasn't lucky consulting the snom docs, maybe the snom support > can be helpful with information about the exact implementation > details. > > You also could use "sip

Am 05.03.2015 um 01:09 schrieb James B. Byrne: > I am trying to determine how the transfer button on the Snom-870 works > with Asterisk. Is the ## special code employed as when it is entered > through the handset or is the blind transfer through the phone > function accomplished in a different fashion? > Hi, I hope I understood your question correctly. AFAIK, the transfer

Check the Redirection on the web interface of your Snom 200. If it says "When Busy" that's your problem. It should say "Never". Also make sure on Sip->Lines your line appearance says "All" or else you will have the same problem. Hope this helps :) Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original

It's very possible that the Polycom IP600 will work with this. As it is just an implementation of a SIP standard for subscribing to the state of other extensions. As for the feature improvements you might see them from me, but not very likely. It is easier for me to train my customers to hit *8 (I will probably just program a pickup button for them) than it is for me to figure out what I

I am trying to get one of the function keys on the Snom 200 working as an intercom. However, I can't get the other Snom 200 phone to auto-answer. I found some posts in the archives from Christian that talk about intercom=true and also the Call-Info header. However, I can't get either one to work. I have tried firmware 2.04g,2.04h,2.05f,3.33 and none work. I am using chan_sip2z.c and

Some switches provide the functionality to try a number till it becomes available. Thus when one dials a number and get a busy, one enters a *XX# code and the switch will call your extension when the called party becomes available. Has somebody already built this in/for Asterisk, otherwise I'll look into it. Michiel Betel Consultancy Abelenlaan 19 T: +31 20 640 3018 1185 RT

Hey, We have asterisk 1.2.4 (old I know) with a couple of snom phones, a couple of grandstream phones and around 65 philips dect stations. Now the problem: All calls do peer to peer RTP except the calls from dect station to dect station. snom to dect or dect to snom do peer to peer. So the sip config looks fine because all the 'static deskphones' honor the REINVITE and start talking to

Hi, We are using threewaycalling & flash transfers over a CAC channelbank. The following happens: Call comes in to my extension I talk to a party and press flash party goes on hold, I get get dail tone I dial internal number internal party answers I press flash once more we are now in a three party conference Or I hang up, and thus transfer the call. Thats fine, but.... What if the