search for: sip_phon

Hi, Is there a way to send and receive SMS over SIP protocol with Asterisk ? I mean, between two SIP phones like below... SIP_phone "A" (sending sms) ====> Asterisk ========>SIP_phone "B" (receiving sms) ... Is it possible ? If so, how could I do it ? Thanks, Angel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/att...

Hi! I've configured LDAP to read both users and extensions from LDAP server. However, I'm experiencing problems with state tracking. Previously when using static files, I was able to map extension number with channel state using: [sip_phones] .... exten => 100,hint,SIP/user exten => user,hint,SIP/user .. rest of the dialplan ... Thus when someone called the user, hint SIP/user showed channel state as BUSY and I was able to use call limits etc. Now I've added this line to [sip_phones]: switch => Realtime/@ My hints,...

Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-----[*]----[mediant2k]-----[Avaya_PBX]------e1-----[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give dialtone from pstn how to setup extention.conf for outside call -----------...

Hello, I try configuring Quintum DX gateway as link to PSTN for *. Now, I can dial number which is connect to Quintum, and call is diverted to *. I don't know what I should set, if I want call from SIP_phone registred in Asterisk to PSTN via Quitnum. I set in sip.conf account for Quintum [sip_proxy-out] type=peer outboundproxy=QUINTUM_IP , and changed extensions.conf. When I call from SIP Phone, I see in Quintum log, that call is received with good caller and called numbers, but I...

...etworkpts.com' failed for '64.169.xx.yyy' - Username/auth name mismatch What's the right way to do this. Shown below are my configuration files: </edg> -----SIPDefault and SIPMacAddr ----- image_version: P0S3-07-4-00 tftp_cfg_dir: "" ; Example: ./sip_phone/ proxy_register: 1 timer_register_expires: 3600 dial_template: dialplan messages_uri: 3688 telnet_level: 2 phone_label: "Nat One" line1_name: 3115552368 line1_shortname: 3115552368 line1_authname: 3115552368 line1_password: 3115552368 line1_displayname: "Nat One" logo_url: &quo...

Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details you can find it here http://mike.calle69.net/IPC5000-brochure.PDF, and its almost the same price as Wisip. But when I ask if this phone will work with asterisk I got this answer "We didn't test...

...;Bad Request" back from 172.16.200.32 And then the phone rejects the call based on a malformed From: header. The from-pstn context is really simple at this point: [from-pstn] ; CALLS FROM PSTN NETWORK ; ;exten => s,1,VoicemailMain2 exten => s,1,NoOp(${CALLERID}) exten => s,2,Dial(${SIP_PHONE},15,j) exten => s,3,Voicemail(u1000) ; Gavin's voicemail, unavailable exten => s,4,Hangup exten => s,103,Voicemail(b1000) ; On the phone exten => s,104,Hangup My Zapata.conf file hasn't changed since 1.0.x and looks like the following: [channels] ; entries...

...e goes in some kind of timeout wait... who knows... Does anyone else experienced common problems? Any help to resolve the problem will be appreciated And here are my .conf files ... very basic yet ;-) ==== extensions.conf === [general] static=yes writeprotect=no [globals] SIP_XLITE = SIP/xlite SIP_PHONE = SIP/sipphone H323_SJPHONE = H323/sjphone@192.168.0.1 H323_PHONE = H323/h323phone@192.168.0.101 IAX_FIREFLY = IAX2/firefly ; ; Inbound ; [inbound] exten => s, 1, Answer exten => s, 2, Playback(ss-noservice) exten => s, 3, Hangup ; ; Internal Extensions ; [local] exten => 10,1,Di...

I write the email again, the third time!!, cause the other two ones, I have had problems while sending them. I hope this time it works. Here is the email again: Hi (and sorry) everybody I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have

Dear sir I have setup Avaya with mediant with asterisk [sip_phone]---[ * ]---[mediant]---E1-trunk--[Avaya]---[analog_phone] This is my configuration when i call from SIP phone i got ringing sound of phone but whn i call from analog_phone behind avaya i didn't get ring sound of ring but SIP phone speaker ring why i am not able to hear ring sound from analog...

I see my own number(or remote called num) instead of caller id on incoming calls on my BT-102. but on Xlite everyything is OK. I'm using * latest CVS. - shabanip

Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,

...c sip_retx: 10 ; Default 10 sip_invite_retx: 6 ; Default 6 timer_invite_expires: 180 ; Default 180 sec # Dialplan template (.xml format file relative to the TFTP root directory) dial_template: dialplan # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: "" ; Example: ./sip_phone/ # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: "137.222.10.60" ; SNTP Server IP Address sntp_mode: anycast ; unicast, multicast, anycast, or directedbroadcast (default) time_zone: GMT ; Time Zone Phone is in dst_offset:...

...imer_invite_expires: 180 ; Default 180 sec ####### New Parameters added in Release 2.0 ####### # Dialplan template (.xml format file relative to the TFTP root directory) dial_template: dialplan # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: "" ; Example: ./sip_phone/ # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: "192.168.17.11" ; SNTP Server IP Address sntp_mode: directedbroadcast ; unicast, multicast, anycast, or directedbroadcast (default) time_zone: YST ; Time Zon...

Hi everyone, I've been toying with * for quite some time now. I've got two Cisco 7940's with the SIP firmware playing nice with *. I can also make outbound calls via IAXTel (toll-free calls only) and all other calls I have routed out my X100P-clone adapter. Here's my question... Is there a way to capture the inbound callerid from my phone line (coming in on the X100P) and have