similar to: How many line appearance can Snom 200 handle?

I have the dtmfmode in sip.conf set to use rfc 2833 however, when my users have to enter pin numbers to join let say someone's conference bridge the pin is received twice. Any ideas on how to solve this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050120/56ff46fa/attachment.htm

I am just playing with a SNOM 190. Overall, I'm very impressed with the quality of the unit and the feature set. I am running the latest firmware (snom190-SIP 3.57u) and the asterisk CVS from last night (1/3/05). The only problem that I've encountered so far is with Call Forwarding, which doesn't work at all. The Snom phone is sending a "486 - Busy Here" back to *, which

Anyone know how I can push a firmware update to a Snom 190 without using DHCP? In the web interface, I specify a path to the Snom firmware, and it works, except I have to physically press OK to get the update to download. I need to do it remotely...

Good day all I have asterisk on a box with one network card I have a 2 companies setup on the system. To keep all apart I binded a different ip to the interface,i,o,w eth0 192.168.0.254 and eth0:1 192.168.1.254 And in sip.conf I took the bind setting out So each company's phones are on a different ip range,and all worked well So we decide to pull the snom190 out and exchange it with a snom360

Hello, I have a customer who said that their Snom 360 is joining calls by accident. The situation is that they had one call on the line and another call came in. She pressed the hold button on the phone and the two calls were joined together. I do have "Call join on Xfer" set to yes, but I thought that would only come into play when doing a transfer, not putting someone on hold. The

Chris Albertson wrote: |What firmware version do you have? program version 1.0.4.39 -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url :

In relation to voice degradation when having 2 or more connection to Asterisk. The comment on the network setup is quite possible. I am not too familiar with linux. How do I check whether the asterisk server's nic is running at full-duplex mode. Does Asterisk use the sound card on the box to do voice processing? I am running xlite on 2 pc and making calls through iax, FWD and back to my

Hi, I've tried to use SIPSAK to understand the troubles i'm having about sending my voice to the person I've called (extension), after doing this tests below I always got this error "error: this FQDN or IP is not valid: voicegw". This could cause problems (namely audio problems)? Best regards, Helder voicegw:~# sipsak -C empty -a password -s

For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the volume button but it also control the

Just got GS 101 phone and plugged into the network. Got ip setup however, the following problems arise: 1. when dialing an extension, I cannot further send any key tone to Asterisk. 2. there is no sound coming from the other end. I have a sip.conf setup for GS: [General] disallow=all allow=ulaw allow=alaw [gs] canreinvite=no dtmfmode=info In the GS101 setting rtp port = 5004 sip port = 5060

I have just checked the Openbsd box on the if interface. fxp0: flags=8843<UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST> mtu 1500 address: 00:02:55:30:54:28 media: Ethernet autoselect (100baseTX full-duplex) status: active inet 192.168.1.1 netmask 0xffffff00 broadcast 192.168.1.255 inet6 fe80::202:55ff:fe30:5428%fxp0 prefixlen 64 scopeid 0x1 xl0:

I am running v2.0.7 and have set up a network drive for an accounting ledger system. The software is called MYOB and is quite popular in Australia. This is the first time I have to deal with multi user access to 1 data file. My setup is: Global oplocks = yes socket options = TCP_NODELAY socket options = IPTOS_LOWDELAY [MYOB] path=/home/office/MYOB force group = office directory mask = 0770

OK, last showstopper that I just can't puzzle my way through - parking calls with the snom phones. I get the two phones connected, I hit transfer on one, the other phone goes to MOH and the first phone gives me DT, so I dial 700 and hit the OK button. Call transferred, the SNOM hangs up before I have a chance to hear which extension it parked to. Is there a way to make the SNOM phones

My ADSL connection speed is 512Kb up and 128Kb down. When making calls from Asterisk to IAX and back to the Asterisk, the sound is choppy and 20% of voice messages was lost. What is the production bandwidth requirement per internet call. I understand there is no guarantee of QoS but at least a benchmark to follow. -- David Kwok Iaxtel/FWD # 17001813482 -------------- next part

Does any one regularly reboot GS101? It sometimes lost registration with * and needs to be reboot. What is the best way to do it by cron? David Kwok

In Australia, Telstra, the local telco provides isdn modem for isdn connection. The modem has 2 analogue telephone jacks and a serial port for connection to dialup internet. My question is that will it be possible to use Zaptel TDM02B to connect to the analogue jack instead of getting a fritz card to do the telephony. Will there be less feature if doing so? -- David Kwok, CISSP Tel: 612

I need clarification as to DID in T1 connection. T1 provides 24 channels for voice/data. Do it assign each channel to particular DID. Or you can have unlimited DID to share the 24 channel as an example. ie. Outgoing/incoming traffic is not bound to particular channel. Whatever is available will be used according to the grouping in zapata.conf. -- David Kwok, CISSP Tel: 612 82315701 ext 1002

Hello Although not stictly a asterisk issue, any help would be apreciated. Firstly a few notes on the snom 360, which I have had on a test bed for the last week. Its a great phone, with a good user interface, both physically and its web based one. At its lastest firmware it does have a few quirks, with regards to the way it handles usernames and passwords on the physical interface. These have

I have 1 * box having x100p installed and the other has no zaptel card at all. both of the * box has compiled in ztdummy module and both have been activated by modprobe ztdummy. When using trunk to connect the 2 *box. The one without zaptel card complaint about unable to accept trunked packet: no matching peer. On the one has zaptel card I have tried to remove the ztdummy module and connect

Does anybody know if there is an app that will cause similar to occur on users PC? I have a scenario where users will have snom phones on their desks. Ideally when their phone receives a call I need to popup a web browser with a specific url. Any ideas appreciated. Neil on 5/7/05 10:52 PM, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> wrote: