search for: snom100

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

...a port the phone doesn't seem to be listening on. * is running on 192.168.35.4, the phone is 192.168.35.110. Thanks for any help. === Sip read: LI> INVITE sip:8500@db.intra;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.21.108:5060;branch=z9hG4bK-2xxwsyff0kus Max-Forwards: 70 From: <sip:snom100@db.intra>;tag=yz77r1h8iq To: <sip:8500@db.intra;user=phone> Call-ID: 3eb464dda940-lvrq6eojtnl9@192.168.35.110 CSeq: 1 INVITE Contact: <sip:snom100@192.168.35.110:5060> User-Agent: snom Version 1.15u Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, O...

...essenger 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 codec.:( I wrote to snom if they want to add a feature to be able to choose which gsm I want to use, but no answer yet. Now I'm trying to write you this question:) Can you implement a feature eg. to the sip.conf to be able to define wich codec to use? THX -------------- next...

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 SI...

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

...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 to the internal Snom100 IPadress and a port range forwarding of 16384 - 32768 (UDP) for the RTP traffic. Additionally I tried to change host = dynamic to host = myserver.dyndns.org to ensure the SIP traffic is going to my Linksys ADSL router and be forwarded to the internal SIP 100 phone. But all my effort did not...

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

...re 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 forget to to publish the year, but this is a offline topic. In a first approach I plugged a Snom100 into the network at the satellite hub station. This should simulate a operator telephone in the head quarter.The Snom100 should reach all the clients in the field network behind the VSAT and vice versa. After configuring and rebooting the Snom100 it tried to register with the Asterisk, but th...

...SIP/mack 2 active channel(s) --- and callee disconnect this call, 'show channels' result is following. mack*CLI> show channels Channel (Context Extension Pri ) State Appl. Data 0 active channel(s) but callee still displayed 'Connected with' ( in snom100 case ) and transmit BYE to caller in 'sip debug' result. and next send INVITE by asterisk again in following under. why??? == Spawn extension (default, 110, 1) exited non-zero on 'SIP/mack2-eba6' XXX Need to handle Retransmitting XXX: BYE sip:mack2 at 192.168.0.1 SIP/2.0 Via: SI...

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 hudecof...

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

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 digium web sites, but this i...

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 &quo...

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

Hi, I have just got my SNOM 200 to start doing some real testing with *.. I am trying to use the GSM codec but the quality is really bad, Is that normal? does anyone actually use GSM?? Also are there any 'gotcha's' that I need to look out for so I don't spend hours trying to get somthing working that really doesn't work anyway.. Thanks.. later.. --

Hello, I`m trying to make a call from the snom 100( SIP mode) but whatever number I dial I get a 404 error from Asterisk. Here are my configs and a dump from "sip debug" . But if I make a call from a Zap line (see extension 2382031), it rings the snom phone sip.conf: ------------------------------------------------------------------------------ ; ; SIP Configuration for Asterisk

...read: REGISTER sip:pbx.pronto.tv SIP/2.0 Via: SIP/2.0/UDP 192.168.32.1:5060;branch=z9hG4bK-ludr3d84pxmp Max-Forwards: 70 From: "Ola Sæverås" <sip:ola@pbx.pronto.tv> To: "Ola Sæverås" <sip:ola@pbx.pronto.tv> Call-ID: 0000000d26d7-zbd2qq9z1svm@192.168.32.1 User-Agent: snom100-1.15e CSeq: 28832 REGISTER Contact: <sip:ola@192.168.32.1:5060;line=1>;q=1.0;description="Available" Expires: 60 Content-Length: 0 11 headers, 0 lines Interface is eth0 IP Address is 192.168.0.10 Using latest request as basis request Sending to 192.168.32.1 : 5060 (non-NAT) Transm...