search for: ipphon

..._cancel' makes pointer from integer without a cast speexec.c:149: warning: comparison between pointer and integer speexec.c:150: warning: passing arg 3 of `speex_preprocess' makes pointer from integer without a cast make[3]: *** [speexec.lo] Error 1 make[3]: Leaving directory `/home/umesh/IPPHONE/linphone-1.5.1/mediastreamer2/src' make[2]: *** [all] Error 2 make[2]: Leaving directory `/home/umesh/IPPHONE/linphone-1.5.1/mediastreamer2/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/home/umesh/IPPHONE/linphone-1.5.1/mediastreamer2' make: *** [all] Error 2...

...me the right configuration for it? My actual configuration is: locale ip: 192.168.0.75 subnet mask: 255.255.255.0 router ip: 192.168.0.1 dns: 192.168.0.1 protocol: iax2 service type: common use service marked service address: 192.168.0.1:4569 nat trasversal: disabled phone number: 103 account: ipphone pin: test register port: 4569 signal port:1701 control port:1721 rtp port:1721 local type: auto the phone firmware is V.1.38.009 My Asterisk extension is: [ipphone] type=friend username=ipphone secret=test auth=md5 host=dynamic context=fullaccess mailbox=103 callerid="ipphone"<103&...

Hi, I'm having some problem with realtime: let's say I have a dialplan like this [globals] IPPHONES=_3XX [sip] exten=>${IPPHONES},1,Answer A call from ip phone 300 comes in, and it's been answered. Then I "switch" the sip context to realtime, putting the exten in the db and using the variable name for this as in the file version. [globals] IPPHONES=_3XX [sip] switch=&...

...me didn't save / documented those password. Now the server's hardware is begin to broke, it hangs a lot, and have a lot of call problem. We already have a new asterisk PBX to replace it, but we have difficulty to retrieve the encrypted password. about a hundred of our customer use an old IPPhone that doesn't have a reset button to hard reset the admin password (back to factory default). The previous engineer also change the IPPhone's admin password without any documentation. So, we can not move / change those IPPhone to the new PBX. Is there a way for us to retrieve / decrypt tho...

...elease. ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gareth Owen Sent: 15 February 2006 02:00 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Firmware version 1.3.1 released for Aastra IPphones Aastra Telecom has released SIP v1.3.1 firmware for the Aastra range of IP phones (480i, 480iCT, 9112i and 9133i). The firmware and release notes (no updated admin and user guides yet) are available for download at: http://www.aastra.com/support/enterpriseip Contrary to what the version numb...

Hello, I'm looking for ipPhone and ATA with UPNP and perhaps also STUN auto provisioning via https or . G729 If someone know a good product.. Thanks Laurent -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attach...

...Cory Andrews -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Sent: Tuesday, January 23, 2007 3:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] OT: High Quality Wireless Headset for Cisco IPPhones and * Has anyone found a high quality wireless headset that works well with Cisco 7960 IP phones on an asterisk system? I tried the vxxi offering but the sound quality was pretty bad. Since these are pricey, I don't want to sample blindly. Experience appreciated. Thanks, Tom __________...

Hi Guys, i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way: 1) use a phone in PBX1 2) call extension in PBX2 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a cellphone) my questions now is : am i gonna be able to dial from an IPphone registered within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2? anybody know IPphone------------->PBX1-----IAX-------->PBX2--------PRI line------------------->cellphone??????????? thank you for you help guys!! -- Abdullah -------------- next part -------------- An HTML a...

...my ip phone using Dial application. Everything works fine, except when I pickup the phone, I can talk, the other party can hear me, but I cannot hear anything the person says on the ip phone. Then after a couple of seconds, the call hangs up. I don't know why. Here is the message I get: SIP/ipphone-09401f10 answered SIP/XX.XX.XXX.XX-09400918 -- Native bridging SIP/XX.XX.XXX.XX-09400918 and SIP/ipphone-09401f10 [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2787 retrans_pkt: Maximum retries exceeded on transmission 4056be591b329cc9441f75b4560c3ccb at XX.XX.XXX.XX<4056be591b329cc9441f75b45...

Im having a little problem with voicemail and my cisco phones i was wondering if anyone might have seen this before and let me know whats going on. it spits this out and then my cisco ip phone reboots im using the latest cvs and a cisco 7910 phone WARNING[1234379840]: File res_adsi.c, Line 205 (__adsi_transmit_messages): Unable to send CAS -- Playing 'vm-login' (language 'en')

Hello, I've one astribank with 8 FXO unit and 8 pstn lines connected to the astribank. When I receive calls on my ipphone I get always Unknown callerid. It's is possible to receive the callerid from the lines on the astribank unit? This is my config: [channels] language=es context=from-zaptel signalling=fxs_ks ;rxwink=300 usecallerid=yes callerid=asreceived ;cidsignalling=bell ;cidstart=ring hidecallerid=no cal...

Hi, I tried set up a global var for an extension, like this [globals] IPPHONES=_3XX [sip] exten=>${IPPHONES},1,Answer What I would like to do is to make a dialplan without fixed extension numbers to change the entire dialplan according to the customer requests: what exten number do you want for your IP Phones ? let me change a variable and we are set! It seems that t...

...39;s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio i have scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for interanet no PSTN at all just only IPphones and analog phones connected on FXS port.Is it's neccassary to cannect with PSTN i don't want i'll just want to use my internal/company premesis. ipphones connected through ethernet while analog phones directly connected through FXS port is that possible i integrate Tenor AXT 800 in su...

...ho_cancel' makes pointer from integer without a cast speexec.c:149: warning: comparison between pointer and integer speexec.c:150: warning: passing arg 3 of `speex_preprocess' makes pointer from integer without a cast make[3]: *** [speexec.lo] Error 1 make[3]: Leaving directory `/home/umesh/IPPHONE/linphone-1.5.1/mediastreamer2/src' make[2]: *** [all] Error 2 make[2]: Leaving directory `/home/umesh/IPPHONE/linphone-1.5.1/mediastreamer2/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/home/umesh/IPPHONE/linphone-1.5.1/mediastreamer2' make: *** [all] Error 2...

I don't know if this is already common knowledge, and it's not specificly for Asterisk, but if you are using Cisco phones and want to roll XML applications, make sure you have "Connection: close" in your HTTP header. Without it screen loads are very sluggish. In PHP, do: header("Connection: close"); I whipped up quick-and-dirty PHP/MySQL/Cisco XML directory and

Hi all, I am trying to use a "Siemens optiPoint 300" IPPhone (H.323 only) with Asterisk (1.0-RC2). So far I have been using the H.323 channel included in the tarball (Nufone ?). I encountered a strange behaviour when I try to make a call from the IPPhone to my Asterisk box : =====> here is the H.323 configuration for the incoming calls (192.16...

Aloha, Has anyone written an XML application for a Ciso 7960 phone running SIP? I can't find any examples anywhere! Anyone know of any resources for this? I have read it can render XML & can get input from the keypad & softkeys. Aloha, Matt

...I tried x-lite on my notebook with wireless connection(802.11). The software has been tested with the fixed line connection. It worked fine to call through *. When using wireless connection, it is clear on my side using notebook; however, there is loud noise on the other side of the call which uses IPphone. It seems to me that some interference noise comes into the upstream. Does anyone notice the same problem? or have explanation of the cause? regards, steven

...oise' undeclared (first use in this function) speexec.c:112: parse error before ')' token cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' make[3]: *** [speexec.lo] Error 1 make[3]: Leaving directory `/home/umesh/IPPHONE/linphone-1.5.1/mediastreamer2/src' make[2]: *** [all] Error 2 make[2]: Leaving directory `/home/umesh/IPPHONE/linphone-1.5.1/mediastreamer2/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/home/umesh/IPPHONE/linphone-1.5.1/mediastreamer2' make: *** [all] Error 2...

Dear all, I have PBX with asterisk 13.x a couple of IPPhone that connect to that asterisk PBX send an alphanumeric dialed phone number. for example, in my CDR table, field DST, it show dialed phone number like - 0C81318304632C (it should be 081318304632) - 08D11157112 (it should be 0811157112). Why it's happening ? and how can I prevent it to happen...