similar to: Failed calls from SNOM100 to *

I'm trying to get a Snom100 configured with H.323. Right now, the phone is not even connecting to the Asterisk server, so there's obviously a problem with the snom config. Does anybody have a sample working configuration with the snom phone, using H.323? I've checked the archives, but everybody seems to be using SIP with the Snom phone, not H.323. -Tilghman

OK I did some researches and tests with it, and finally: I registered my messenger to the asterisk and called if from the snom. The audio from the snom to the messenger was PERFECT. By the time of the call This message was running on the asterisk console: WARNING[16400]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames My conclusion is that the snom100 utilizes MSGSM

Hi, there were some postings a few weeks ago telling that the GSM codec problem with snom100 will be fixed. But it still seems to be very quality. Will be any change in this subject? THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/2fa9d206/attachment.htm

Hi, Just wondering if anyone else is able to reproduce this with the current * (CVS 12:15 GMT) Call Snom from any device (tested with i4l, zaptel and SIP). Answer call and try to transfer call using transfer button on Snom. After dialing new number press OK (F4). At this moment the Snom users hears dialtone, but the caller still hears the Snom user... Even hangup on the Snom doesn't

Hi all, -------- I have a Asterisk at a public Network (official IP address). In the local network I have isntalled a Snom 200 IP phone and in my home network (behind NAT) a Snom 100 device. I can dial the Snom200 device from my home location without any problems but the Snom200 can not dial me. It always gets a "we do not rely". I tried to forward the SIP Port (5060) UDP via UPnP

Hi all, ------ I tried to use * over satellite, but all my effort did not succeed. The Asterisk is behind the VSAT and is resposibel for alle the SIP clients in a field location. The clients are notebooks and PDA's running SJPhoen for Windows and PocketPC. Unfortunately I could not find any Linux Client wich worked satisfying. SJ LAbs promised a Linux Version at the end of August but they

sorry, this problem is fixed by myself. we must need set 'canreinvite=no' each user. --- I'm try to discconect a call with SIP. when caller make a call, 'show channels' result is following. mack*CLI> show channels Channel (Context Extension Pri ) State Appl. Data SIP/mack-1bfc (default 1 ) Ringing AppDial (Outgoing

hello, can anybody help me with folloving problem I have asterisk with the public IP and two UAs (snom100, x-lite) in the same private network behind the same FW/NAT. All is working good, but whan I tried to establish call between these two UAs, first 10-15 second is nothing to hear and then is the quality terrible :( Can anyone tell how to get it work with normal quality ? best regards

Hi all, I have been struggling desperately to get * work together with my snom100 for days on end, but I am not making any progress... Of the entries marked *#) I'm still not sure what it does; so far I have on the snom in "SIP/lines" -user name - empty *1) -account - Conrad -registrar - 192.168.200.83 -action - "None" *2) in

Hello. I've been going through the archives, but can't discern the state or future direction of IAX on the SNOM100. The most recent image appears to be from September 2002. There was a message on this list stating that SNOM was coming to visit Digium last April with the intention of adding IAX support themselves. For a while there was reference to the I100E on the asterisk and/or

On Wed, Dec 10, 2014 at 2:03 PM, Dan Cropp <dan at amtelco.com> wrote: > Thanks George. > > > > That was the ip address I was given. Unfortunately, my contact at > Vitelity is gone for the day so I can?t verify it with him. > > > > I added the qualify_frequency as you suggested and it does appear that I > have something configured incorrectly?. > >

hi, can some one tell me a good IP phone (not software, but a "real" phone :) that work well with asterisk? how mutch does it cost a good IP phone? i made a VoIP network for my company, but now we are using a client for PC phone... i'd like to buy a IP phone, can someone tell me witch model i should buy? thanks, Angelo

Thanks George. That was the ip address I was given. Unfortunately, my contact at Vitelity is gone for the day so I can?t verify it with him. I added the qualify_frequency as you suggested and it does appear that I have something configured incorrectly?. <--- Transmitting SIP request (483 bytes) to UDP:0.0.19.196:5060 ---> OPTIONS sip:64.2.142.93 at 5060 SIP/2.0 Via: SIP/2.0/UDP

hi all when starting kphone, it tries to register with asterisk but fails after a while. The SIP entry in * for this user is below. This is identical to the other SIP entries. The other SIP clients are MSN messenger plus one snom. these work fine. See SIP debug output attached as 'screen-exchange' thanks roy [roy] type=friend ;insecure=yes username=roy ;secret=password host=dynamic

hi My stations are behinds a firewall, the system is windows 2000 & 98, i use sjphone asterisk is on the internet gateway where is the firewall Shorewall the system is linux debian (sid) kernel 2.4.20 j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy) to write my sip.conf but i can't call an external sip user. (an external user can call me) i try without asterisk with

On 22/11/15 09:14, Andrey Repin wrote: > Greetings, Rowland Penny! > >>>>> Is there at last a solution? I've only found questions, in the list, and on >>>>> the network. >>>>> >>>>> The issue is that DC built on Samba4 does not report to network browsers >>>>> neither it is participating in election to become

Hi folks, in order to establish backward compatibility we made an image that automatically detects if the other side does not support RFC3264. Please try it out, we would be very interested if this image is a progress! http://snom.com/download/share/snom200-2.03b-SIP.bin Thanks, CS

Hi all! Please I need someone that have already done an H.323 - SIP gateway to help me with some problems. I can stablish calls from a SIP telephone to a H.323, but I can't do vice versa... (problems with port 1719- when the gatekeeper tries to contact with asterisk at this port, it is unrecheable...). Please someone can help me? Regards, Mireia

hi all anyone that've heard of any working IAX hardphones yet? roy

Hi all I have 2 snom 100's and an ix66 (sip aware firewall) set up with asterisk. I needed to register a number of lines so what I've done is make asterisk register all the lines i need (attaching them to an extention eg 1000) and then register each phone with asterisk. so for example in sip.conf: register => andy@sip.mydomain1.org/1000 register => andy@sip.mydomain2.org/1000