search for: 79x0

...key, the name is correct, and the Phone field shows "asterisk", or whatever it was changed to by the fromuser setting. I am just curious if this is something anyone has thought about. Is there anyway there could be a specific command to change SIP header when a Dial command is sent to a 79x0 phone to reflect the CID number in the From: command. I don't know if the cisco phones are the only ones that have this issue. Also, I read a message on this list from September that asked this same basic question. The answer given was to make sure the CID is correct. I have tried many combi...

Hi, I've just found: http://lists.digium.com/pipermail/asterisk-users/2003-June/014475.html which talks about ALERT_INFO and Cisco phones. How do I actually get this working and what does it do? Do I need to add anything to the configs for the phone or is it just a SetVar(ALERT_INFO=1) - which I tried and it seemed to do nothing at all.. Thanks Andy

Cisco has version 7.1 of their SIP firmware for the 79x0 phones. They advertise "no new software features", but it does include bugfixes for a number of things. I know there was a discussion about the 0.4sec delay, which is said to be resolved in this firmware (CSCed48311: Media takes 0.4 sec to be set up) Steve

Hi list, I have bought some Cisco 7941G-GE IP phones and want to use them with asterisk. Before bying I tested the whole setup with three different models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the formerly provided SCCP-Image to SIP was no problem, but now it complains about a nonexistent CTLSEP<mac>.tlv file. Most of the howtos say something about an empty file but that does not suit to me. Does anyone of you have experience in getting th...

Hello, I have read the information on echo and SIP in the FAQ and I have scoured the mailing list for possible solutions, but as yet I have not been able to get rid of this echo. I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed into an asterisk server. If I call between the Sip Phone (Budgettone-100) and the 4 FXS ports everything sounds great. If I call out to the PSTN

...r some of the topics below..... Thanx in advance for any help. Chris. * We currently have a Cisco IP telephony system (using their CallManager).....am I right in saying that Cisco phones using Skinny will not work with asterisk? Is it ever likely too? * When we connect and power on a Cisco 79X0 phone for the first time, it automatically registers with the CallManager and is assigned a temporary number. We then do into the CallManager admin interface and assign it to its owner, give it its permanent number etc. Among the things which happen are the TFTP files for the phone (eg: SEP<...

I am guessing you are running without reinvite's, I'm running with reinvite's with latest CVS release and 79x0 phones without any issues with conferencing... > -----Original Message----- > From: Adam Rothschild [mailto:asr@latency.net] > Sent: 08 October 2003 15:49 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Cisco 7940/7960 phone and > conference calling? > &gt...

Hi, (I think) I followed instructions here ( http://www.voip-info.org/wiki/view/Firmware+issues+on+7940+-+7960 section "Notes added Nov 2005, revised May 2006:" at the bottom of the page) to factory reset a Cisco 7942 I wanted to configure to SIP firmware. When booting, I can see this requesting and obtaining file term42.default.loads from TFTP server. Then it would send a request

Hello, LLDP is more and more available on various network elements (endpoint, switches, ...). It seems to ease network configuration. Do you have any experience with it ? How would you rate LLDP ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091124/fce6307c/attachment.htm

I'm currently running firmware version 3.2 on my Cisco 7960. I've seen on the list that several people are running the 5.x latest versions. I've avoided going to higher firmware versions because I'm worried about potential problems or issues with the encryption mechanism used in the later firmware versions. (Once you go to an encrypted firmware version, you can't go back,

For anyone who is interested, I have a working tftpd (modified wvtftpd) capable of serving configuration, dialplans, and ringtones to Cisco 7960/7940 and ATA-186 devices that are located behind NAT firewalls. As TFTP is not a very firewall/NAT friendly protocol, I had to break some rules to get it to work with these cisco devices. It might cause problems for other TFTP clients, but it works with

For those that might be using Cisco 7940/7960 sip phones and placing calls across an iax2 link, we think the voice quality problem has been identified and corrected. The dev cvs should be updated as of about 3:30pm CDT today (April 14). History: Calls originating from a Cisco 79x0 sip phone and sent via iax2 link to some distant * machine resulted in very poor quality audio, and in some cases, the audio was so choppy as to be unusable. The quality problem surfaced around March 5th when coding changes were made to directly associate iax2 timestamps with the sip/rtp timestam...

I have a client that has an old Merlin system. They would like to move to an Asterisk based system, however, with their existing system each phone is capable of displaying who is on the phone within there office. This is done by lighting a red light for each line(extension) that is in use. Has anyone been able to neatly create this feature? Perhaps an XML application can be written for the Cisco

hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing "not registered". i think asterisk is properly sending request to UA. any comments............this sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved

...> > PC's network as well if it is plugged into the phone. > > Question 2 I would like to know the answer to myself. I would be > > curious to know if it works with the SIP image in call manager. > > Same here. > > We have about 500 phones, from both 79x1 and 79x0 series; > I posted the same two questions twice some time ago but never > got an answer: I do reboot phones by power cycling them too, > while I've been able to use blf with sccp images only. > > Furthermore, XML Services on 7940/7960 seem to be broken > or at least to behave...

Hi, I want to know, if someone has tried to use clustering in asterisk to increase its scalability and make it distributed?? If yes, how easy it is to cluster? Can someone please ive me details about the same Thanks Varun __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail

Hi all, I'm looking for recommendations on ethernet switches for a new install. Ideally would want switches with at least 24 ports, ideally with a GE uplink, and that support PoE (power over ethernet) on every port. I've seen lots of switches, and lots of "power hubs", but the combination, which makes a lot of sense, seems rare. What is out there? Do the switches need to be

I run a SIP-only shop with a 23 channel PRI and single T100P. Here are my configs: /etc/zaptel.conf: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us /etc/asterisk/zapata.conf [channels] context=default switchtype=dms100 signalling=pri_cpe pridialplan=unknown rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=no hidecallerid=no callwaiting=no

...gt; > > > > > I have bought some Cisco 7941G-GE IP phones and want to use > > > > > them with > > > > > > asterisk. Before bying I tested the whole setup with three > > > > > different > > > > > > models of the old 79X0 series (a 7912, 7940 and a 7960). > > > > > Flashing the > > > > > > formerly provided SCCP-Image to SIP was no problem, but now it > > > > > complains > > > > > > about a nonexistent CTLSEP<mac>.tlv file. Most of the h...