search for: ipdialog

Here is a SIP phone I haven't seen before. Does anyone have any experience with this one? -----Original Message----- From: George Richardson [mailto:georger@netxusa.com] Sent: Wednesday, April 02, 2003 4:56 PM To: clay@ctitec.com Subject: ipDialog Ethernet SIP Phone $199 pad <http://us.st1.yimg.com/store1.yimg.com/Img/trans_1x1.gif> <http://store1.yimg.com/I/sysconfig_1733_51893> <http://store1.yimg.com/I/sysconfig_1733_14909> Click to enlarge$199.00 ipDialog SipTone Ethernet Phone <http://store1.yimg.com...

I just got 2 IpDialog phones for use with my Asterisk system. I have been able to get the phones to just dial local extensions but it is not able to register with my system correctly. I would like to know if someone has set these phones up before and how they did it! Is there any examples for use with Asterisk? They...

Dear Asterisk Group. I have two Asterisk servers serving two data/help desk centers, both centers have a near identical setup. However, when connected to one of my data centers, I call a user, I can see on the CLI that the phone is ringing, but I hear no ringing on my SIP soft phone? Has anyone had a similar scenario? How as it resolved. Warm Regards Shad Mortazavi

Hello, I would like to buy a SIP IP phone, but I don't know wich one to choose... Can you tell me wich IP Phone is known to work with Asterisk please. I've seen the Cisco 7940, but I don't know if it works, and how expensive is it ? I'm french, so if you know some french resellers, tell me. Thanks a lot, ---------------------- Fabrice Tereszkiewicz Sawadka.org

...5.03 that I am using. Cisco Support said I should not upgrade to version 6 yet. My next question is the sound is patchy when people here me. But I can hear them just fine not patchy. I have the 188 page Admin manual and it seem not to say anything about improving the sound. All other phones like IPDialog work fine without the patchy sound. I have tried ulaw and alaw as the codex. Both sound the same! Is there any other settings that can be done. I remember that the X-lite has a transmit silence but I could not find this setting in there documentation. P.S. the Contract for the 7960 cost us $...

...h Asterisk. I did not see this problem as having a fix or having a patch for it. I can not use the # in our system due to IVR systems we access. Can someone let me know at what stage this is at. This is a major problem with our system in deploying SIP phones. We have Cisco 7960, Snom 200 and IpDialog's working but can not transfer. Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040130/929453a0/attachment.htm

...o is to have * prefix the CID External (so that I can tell that it is a fresh call) or Internal (to tell me that it is a blind forward). So how do I prefix the CID with say E for external and I for Internal? We are using an AVM C2 ISDN card and are in the UK (if any of that helps?) Our phones are ipDialog SipTone II Many thanks in advance and kind regards to all. Nick

...If I initiate the call from the siptone and the other end hangs up first, same thing -- congestion. The same thing happens if we make calls from the analog phones attached to the Mediatrix 1102. This does not happen on our Snom 200 phones, which have similar entries in the Sip.conf as the the IpDialog. I don't know what it could be considering it is on two completely different phone technologies. Any help would be greatly appreciated. We are running the latest CVS build. Thanks in advance. exten => s,1,Dial(${ARG1},20,tr) exten => s,2,Voicemail(u${MACRO_EXTEN}) exten =&...

...Buffalo, NY 14225 (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM: B2sales Specializing in New and Used equipment from vendors including Cisco Systems, Juniper, Adtran, Dialogic, Lucent, Nortel, Sipura, Granstream, Snom, Mediatrix, Carrier Access, Digium, Zultys, IPDialog and more. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041123/8831fec2/attachment.htm

I have a Cisco 7960 & IpDialogs that I am not able to use the transfer button on it. What happens is that it puts the call on hold and then it gives you a dial tone. You can dial but it will not transfer the call. What we are trying to do is transfer to extension 700 for parking so another person can pick up the line. We can...

Hi! I am doing a research about the prices of SIP telephones. If someone can tell me which one are the cheapest and have an acceptable quality... it will be very kind. Best Regards, Mireia

Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike

...10431] chan_sip.c: -REALTIME- loading peer from database to memory. Name: stone. Peer objects: 8 [2009-09-23 11:10:57.3q] VERBOSE[10431] chan_sip.c: -- Registered SIP 'stone' at 10.215.42.138 port 5060 [2009-09-23 11:10:57.3q] VERBOSE[10431] chan_sip.c: > Saved useragent "ipDialog SipTone 1.2.0 rc Z_21 UA" for peer stone [2009-09-23 11:10:57.3q] WARNING[10431] res_config_odbc.c: Key field 'ipaddr' does not exist in table 'foo.sip at asterisk'. Update will fail [2009-09-23 11:10:57.3q] DEBUG[10431] res_config_odbc.c: Skip: 62; SQL: UPDATE public.sip SET...

...4. PSTN Connectivity a. Digium Hardware i. X100P ii. TDM400P iii. T100P iv. TE405P v. IAXy b. Other Hardware i. Sippura SPA-2000 and 3000 ii. Virbiage IAX2 ATA iii. Cisco ATA 186 c. SIP Phones i. Cisco 7905, 7912, 7940 and 7960 ii. SNOM 190, 200 and 220 (Running Linux) iii. ipDialog SIP II Phone d. ITSP Providers i. N2Net Univoice (IAX2 / SIP) ii. NuFone (IAX2 / SIP / H323) iii. Broadvoice (IAX2 / SIP) iv. VoicePulse (SIP) 5. Asterisk In The Real World a. Who Uses Asterisk b. Asterisk Installation Examples 6. Where Can I Get More Information a. Asterisk Website = h...

...kup parked calls and to park them. Most of them are on firmware 2.04g I have upgraded one to 2.05c for testing but this did not fix the problem. I get no error message on the CLI and I am at a lost of where I can begin to look for a problem. I have other Sip phones working fine. Cisco 7960'g, IpDialogs They all work fine. ATA 186 and Sipura-2000 are also working fine they all can park a call and pick them up. ----------------------------------------------------------------- \ \____\_ Ariel Batista / / / Avionica, Inc. ----------------------------------------...

Hi All, * is running a dream now, however we have an odd problem that I am sure some guru will be able to sort out for me in no time!! When receiving or making a call about 60 seconds or so into the call we develop choppy/stutter audio problems. It then seems to clear itself only to return again, and so the pattern carries on! This has got me stumped! Our equipment is SipTone II handsets, AVM

I am using a IPDialog siptone II. I can take incoming calls, but when I try and make an outgoing call I get a SIP 407 error. Can some kind soul explain to me what I am doing wrong? Here's what I found in the wiki: If a proxy does not accept the credentials sent with a request, it SHOULD return a 407 (Proxy Au...

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Dear List, Can anyone recommend a sip phone for residential use? (asterisk home pbx) Thanks!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040819/7b107ebc/attachment.htm

...Buffalo, NY 14225 (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM: B2sales Specializing in New and Used equipment from vendors including Cisco Systems, Juniper, Adtran, Dialogic, Lucent, Nortel, Sipura, Granstream, Snom, Mediatrix, Carrier Access, Digium, Zultys, IPDialog and more. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041208/081d90f5/attachment.htm