search for: spa504g

Hello, I am struggling to have a SPA504G to auto answer (for intercom/paging). I have tried the following SIP headers (not all together), but without luck: SIPAddHeader(Call-Info:\;answer-after=0); SIPAddHeader(Call-Info: answer-after=0); SIPAddHeader(Alert-Info: info=intercom); SIPAddHeader(Alert-Info: Ring Answer); SIPAddHeader(P-Auto-...

If you have experience with these phones... We are trying to figure out how to transfer an established call on the SPA504G while a second call is incoming. At present, the receptionist has to answer every single incoming call before the XFER softkey is seen again. This is completely unpractical for a receptionist that may have 4 or more calls coming in at the same time. When the receptionist has an established...

exten => 179,1,SIPAddHeader(Call-Info:\;answer-after=0) exten => 179,2,Page(SIP/180&SIP/181&SIP/182&SIP/184) The asterisk11 page() application works great, but I've just learned that the person who initiated the page can transfer or conference the page if they don't hang it up before using those functions. It never would have occurred to me to try it, but a

...n cope with it longer term. Our test involves about 10 BLF-NOTIFY messages per second to each handset with a 5-second pause every 5 seconds. This will either crash or render unusable all of the following combinations: snom360 + 1 x sidecar Yealink T28 + 1 x sidecar Yealink T28 + 2 x sidecar Cisco SPA504g + 1 x sidecar Cisco SPA504g + 2 x sidecar Cisco SPA525g + 1 x sidecar (reboots often) Cisco SPA525g + 2 x sidecar (reboots quickly) Aastra 55i + non-LCD sidecar Did not try Polycom as they do not do directed pickup and only small sidecars. Linksys SPA962 with one sidecar is "OK" but is d...

I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001 at 192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt t...

Hi all, I have to install 25 IP Phone in some building. I want just basic IP Phones like: Cisco-Linksys SPA922 u$s 146 Grandstream GXP-2000 u$s 105 Snom 300 u$s 119 The most valuables parameters for me are (in importance order from high to low): - Stability (device don't hang in any way) - Voice quality using G729 - Provisioning So what device do

Hi, Is it working for anyone? I have tried with trustrpid=yes sendrpid=yes/pai but can not get it working, Asterisk cli shows prevented message like this. Connected line update to SIP/1231-00000200 prevented Regards, Zohair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL:

...to office so phone works just fine. Another user uses VPN and soft phone and it works good too. Now I need to setup some users with actual phone devices and none of those solutions will work. So, I did some research and found that some phones have VPN capability built in. Right now I use Cisco SPA504G phones. We have auto-provisioning for them, works well. But I don?t think they have VPN capability. What I found it that Cisco 525g2 has AnyConnect functionality (SSL VPN) but not sure if this is what I need. We have Mikrotik router. Can I setup VPN on router and have this Cisco phone auto-dial...

Hello, This is not technical post, just looking for suggestions on what to check.I have asterisk for long time, no updates, just maintain OS updates. I use SPA504G phones Very rarely and randomly when we pickup a phone - other side does not hear us. Call them back and all works. Now I have couple people I'm talking to and it seems like very call like this. Someone can't hear someone. Don't know where to start to troubleshoot and what to look for....

I'm trying to implement call parking with asterisk and Cisco SPA504G phones: features.conf parkext => 700 parkpos => 701-702 context => parkedcalls I defined one of the unused keys to park the calls: Key2: fnc=sd;ext=700 at 10.0.1.103;vid=1;nme=Park I also defined two other keys to pickup/unpark the calls: Key3: fnc=blf+sd+cp;sub=701 at 10.0.1.103 Ke...

...i="sip:333 at xxx.xxx.xxx.xxx",algorithm=MD5,response="8dd827e9910c2446fb0b8497f5944b81",opaque="66e52\ 68a2111e777",qop=auth,nc=00000001,cnonce="9dda9e0d"^M Contact: "1004" <sip:1004 at yyy.yyy.yyy.yyy:5063>^M Expires: 240^M User-Agent: Cisco/SPA504G-7.4.8a^M Content-Length: 401^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE^M Supported: replaces^M Content-Type: application/sdp^M ^M v=0^M o=- 32730859 32730859 IN IP4 yyy.yyy.yyy.yyy^M s=-^M c=IN IP4 yyy.yyy.yyy.yyy^M t=0 0^M m=audio 16436 RTP/AVP 0 2 8 9 18 96 97 98 101...

We are planning to have about 100+ parking lots defined in features.conf , each with about 4 unique park positions. Asterisk will be handling all the parking and unparking (we don't exclusively use Park/ParkedCall in the dialplan): [parkinglot_a] parkpos => 1-4 context=parked [parkinglot_b] parkpos => 5-8 context=parked As far as I can tell, Asterisk adds/removes extensions to the

Hello, ? (using trixbox with Asterisk 1.6.0.26) ? I am looking for information about how Asterisk should notify the unavailable (SIP) state of a SIP device. ? I found out that the phone (SPA504G with attendant console) sends a SUBSCRIBE request with an Accept: application/dialog-info+xml. ? The situation is that the BLF leds are green even for phones that are currently not "online". This leads to wrong call transfers from the attendant console into not connected phones by an ope...

I am running asterisk 13.1.0 In pjsip.conf, the endpoint section has an aors and an auth field. I can name the auth field anything I want. The key is to set the auth=field accordingly. However, when I try this with the aors field, it never works. It seems I have to name the aors=field to match the name of the endpoint section. Is this correct? Would there ever be a need for multiple aors to