search for: phone2

...two snom 300 phones to support only g722 and asterisk declined the sip invitation. That is bad for me. Is it possible that asterisk asks the called phone not the codecs that asterisk supports but the codecs that the calling phone supports? What i want: phone1->asterisk: hello, i'm calling phone2, codecs possible: g722 asterisk->phone2: hello, you have a call from phone1, codecs possible: g722 phone2->asterisk: ok, let it be g722 ... chit chat ... But asterisk does this: phone1->asterisk: hello, i'm calling phone2, codecs possible: g722 asterisk->phone2: hello, you have a...

Hello everybody, I am sorry to bring this up again if this kind of echo issue has ever discussed. Phone2 in below call path experiences quite annoying echo: Phone1 --> FXS (TDM400P) --> Asterisk --> SIP GW --> PSTN --> Phone2 It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echo to maximum with the foll...

hi. i have dialplan with 2 simultaneous calls - dial(sip/phone1&sip/phone2). when i cancel call on phone1 (push "reject" button), the call is still ringing on phone2 can i cancel call on both phones from one place(one phone)? thanks -- --------------------------------------- Marek Cervenka =======================================

I have the following setup: AnalogPhone1--TDM400P-ASTERISK---via SIP---Softswitch--------POTS Phone2 When I call from AnalogPhone1 to Phone2 I hear a ringing tone and all is well. When making a call from Phone2, I get a dial tone but after dialing the number I hear nothing (no ringing tone). On Asterisk console it says that a call is coming in and that it is ringing Zap/2. I can also hear the An...

...er VIA asterisk.. I've added this to sip.conf: [phone1] type=friend host=dynamic defaultip=192.168.1.106 ;username=blah ;secret=blah dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info mailbox=1000 ; Mailbox for message waiting indicator context=sip callerid="Me" <2124> [phone2] type=friend ;secret=blah host=dynamic defaultip=192.168.1.107 dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info mailbox=1000 ; Mailbox for message waiting indicator context=sip callerid="Mini Me" <2123> And in extensions.conf at the very end: [sip] exten => 1,1,Dial(SIP/...

...I call to a softphone with a Grandstream I can pich up the call with the softphone but the Grandstream keeps ringing like on the other site you didn't pick up the phone.(even if you do so) It's the same when I call between two Grandstream phone's. Call from phone1 to phone 2, I pick up phone2 and afther 3 seconds I get congestion tone from both phone's. Info from command *CLI> -- Executing Dial("SIP/phone2-a030a", "sip/phone1") in new stack -- Called phone1 -- SIP/phone1-663a is ringing -- SIP/phone1-663a answered SIP/phone2-a030a -- Attempting native bridge...

Hi, all Can someone tell me where to tell asterisk no to use 127.0.0.1 IP (localhost)? When I am registering with VoIP providers, they get my info as s@127.0.0.1. (This is SIP registration). Also, in SIP logs, when calling I am getting things like this: Executing SetCallerID("SIP/phone2-22c3", ""CID Name" <CIDNUMBER>") > in new stack > -- Executing Dial("SIP/phone2-22c3", "SIP/sipnet/84959741926") in new > stack > We're at 127.0.0.1 port 18900 ANy help is appreciated, Thanks, Rudolf

I've just installed asterisk on a Debian Linux (apt-get it) And i have placed two sip phones in sip.conf and i'm testing parking with them I have phone1-SIP/1000 and phone2-SIP/1007 The following happens if i park from calling party and everything is OK 1. Pickup Phone2 and call to Phone1 2. Talk 3. Phone2 dials #700 and parks the call (it is placed in 701) 4. Phone2 is hangup 5. Pickup Phone2 and dial 701 6. The Phone2 is connected back to phone1 7. everything is OK...

Dear, When I start asterisk -vvvvvvgrc and I ask 'sip show peers', I don't get the ip adress in the 'Host" field. Name = phone1 and phone2 Host=unspecified mask 255.255.255.255 port = 0 status = unmonitored I can ping the two phone's and get a reply (also from the laptop) phone ip adres 192.168.10.12 and 192.168.10.13 (server 192.168.10.11and laptop 192.168.10.14) hardware config: server - phone1 - phone2 - laptop configuratio...

Hello I have one phone (phone1) in one network, the other (phone2) in public network. both can call the other side; phone1 can be heard by phone2, phone2 can't be heard. I don't have NAT set on both end, but I use rtpproxy on SER. Is NAT still necessary to be set on both phones? Thank you! steven

Hi all I have registered my phone1 in asterisk server 192.168.0.9 and phone2 in asterisk server 192.168.0.6. Both are sip phones. i configured the extensions.conf file in both the server. the extensions.conf file on server 192.168.0.9 is exten=>301,1,Dial(SIP/301@192.168.0.6,20,tr) exten=>401,1,Dial(SIP/phone1,20,tr) 301 is the extension number for phone 2 in...

...a context, dial one party and then bridge another party) I thought that the G() flag in the dial application would work. I tried the the following test (continue down a dial plan). One station calls into a context ... in this case, dials '55' to start the extenion exten => 55,1,DBget(PHONE2=demo/phone2) exten => 55,2,Playback(/recordings/prompt01) exten => 55,3,Dial(${PHONE2},,rG(from-internal-custom, 55, 4)) exten => 55,4,Playback(/recordings/prompt02) exten => 55,5,Hangup() You would think that the two parties would hear prompt02 and each other in a conversation ... th...

...two snom 300 phones to support only g722 and asterisk declined the sip invitation. That is bad for me. Is it possible that asterisk asks the called phone not the codecs that asterisk supports but the codecs that the calling phone supports? What i want: phone1->asterisk: hello, i'm calling phone2, codecs possible: g722 asterisk->phone2: hello, you have a call from phone1, codecs possible: g722 phone2->asterisk: ok, let it be g722 ... chit chat ... But asterisk does this: phone1->asterisk: hello, i'm calling phone2, codecs possible: g722 asterisk->phone2: hello, you have a...

I don't know if this is possible, and I can't quite get my head around how to do it... If I am using DUNDi for redundancy in a cluster, when Phone1 makes a call to Phone2, both Asterisk A and B will be in the RTP stream: +---+ +---+ | A |-----| B | /+---+ +---+\ / \ Phone1 Phone2 Is there a way configure re-invites in this situation so that either Asterisk A or B drops out of the call, and there&...

...itor(wav49,/var/spool/asterisk/monitor/425/${EPOCH}_${CALLERID(num)}_in,mb) exten => 425,n,Dial(${PHONE1},10) Now, I want to create a call group: I mean, I want a number (eg 800) that makes 3,4...10 phones ringing together. I found 2 modes to do this: 1) exten => 800,n,Dial(${PHONE1}&${PHONE2}&${...},15) 2) with Asterisk 1.6: exten => 800,n,Set(DIALGROUP(test,add)=${PHONE1}) exten => 800,n,Set(DIALGROUP(test,add)=${PHONE2}) .... exten => 800,n,Dial(${DIALGROUP(test)}) How can I record a call made to the number 800 but that will be stored on the directory of the phone (eg...

...um of 4 calls before Asterisk considers the phone busy and don't sends calls to it anymore. The tricky part comes now: the customer asked me to "load balance" the phones, so the next incoming call should be sent to the phone with the least calls. For example: a) Phone1 has one call, Phone2 has two. Next incoming call should be routed to Phone1. b) Phone1 has two calls, Phone2 has zero. Next incoming call should be routed to Phone2. c) Both Phone1 and Phone2 have one call. Next incoming call can be routed to any of them. And so on. I have never done something similar before. I tried...

...ite=no insecure=very and here's the output from sip show peers and sip show registry *CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status grandstream1/grandstream1 (Unspecified) D 255.255.255.255 0 Unmonitored phone2/phone2 (Unspecified) D 255.255.255.255 0 Unmonitored phone1/phone1 192.168.1.108 D 255.255.255.255 5060 Unmonitored simpleconnect-sip/wlee179 63.218.92.199 255.255.255.255 5060 Unmonitored broadvoice2/5083021425...

Now that I am using the chan_sccp module, the phones now work as single line phones. However, these phones have support for multiple lines. So I setup phone1 with extension 1001, and phone2 with exts 1002 and 1003. If I call ext 1003 from 1001, phone2 rings correctly and if I pickup the handset on phone2 I can carry on the conversation. If I call ext 1002 from 1001, phone2 rings as it should and the little icon beside the 1002 ext indicates incomming call as well. But if I pick up t...

...will answer my question. If I have two Asterisk boxes in different locations which are linked to each other over the internet, can I configure the boxes to use each other's lines as local? In other words, let's say Site A has Phone1 for a 1FB line going into it on an FXO port. Site B has Phone2 for a 1FB line going into it on an FXO port. Is there a way to configure Site A to use Phone2 from Site B and vice versa? Undrhil

Hi, Actually on my setup all outgoing calls are going trhu a SIP unique account A have a second SIP account with another operator and I would like my setup to use alternatively each of the two accoutns Call 1=> Dial SIP/phone1 Call 2=> Dial SIP/phone2 Call 3=> Dial SIP/phone1 <...> If you have an sample please let me know