search for: mysip

...Trunk----------[Avaya] When i call from avaya to asterisk i got long ringing tone then hangup but when i call from asterisk to avaya i got 4 ringback and then hangup with this error *CLI> Jun 26 01:26:08 NOTICE[5555]: chan_local.c:523 local_alloc: No such extension/context 1022 at mysip creating local channel Jun 26 01:26:08 NOTICE[5555]: app_dial.c:474 wait_for_answer: Unable to create local channel for call forward to 'Local/1022 at mysip' (cause = 0) My sip.conf [41] type=friend context=mysip username=41 host=dynamic callerid=SIP Phone <41> canreinvit...

...en I tried to make a tcp dump I can see no packets going between the windows client and the Asterisk server at all, here is the my conf on the xlite itself: in the Menu>System Settings>SIP Proxy>Deafult Enabled: yes Display Name: Username: Authorization User: Password: xxxx Domain/Realm: mysip.server.com SIP Proxy: 192.168.99.243 Outbound Proxy: Use Outblound Proxy: Default Send internal IP: Always Register: Always Direct Dial IP: NO DIal Prefix: my sip.conf for the device is as follow: [881] ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! ;Note that Xlite...

...rom the ethernet and the call is uninterruoted. Unfortuanately the call quality is terrible! Low volume, choppy and so on. It seemed to me that since I had stepped my * box out of the network, the problem must be with the ITSP. They suggested I try SIP. With SIP I use: Dial(SIP/7775551234@mySIP) Unfortuantely I don't get the handshakes and the whole call ends up passing through my box, which is something I'm desperate to avoid. I have canreinvite=yes as seen from my sip.conf: [mySIP] type=peer auth=md5 username=<UID> fromuser=<UID> fromdomain=<d...

....77.21.15:11510 Then in extensions.conf I added: exten => 1234,1,NoOp() same => n,Answer() same => n,MusicOnHold(test) same => n,Hangup() CLI> dialplan reload Then I dial: == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 -- Executing [1234 at features:1] NoOp("SIP/mysip_4405-0000001f", "") in new stack -- Executing [1234 at features:2] Answer("SIP/mysip-0000001f", "") in new stack -- Executing [1234 at features:3] MusicOnHold("SIP/mysip-0000001f", "test") in new stack -- Started music on hold, clas...

...rom the ethernet and the call is uninterruoted. Unfortuanately the call quality is terrible! Low volume, choppy and so on. It seemed to me that since I had stepped my * box out of the network, the problem must be with the ITSP. They suggested I try SIP. With SIP I use: Dial(SIP/7775551234@mySIP) Unfortuantely I don't get the handshakes and the whole call ends up passing through my box, which is something I'm desperate to avoid. I have canreinvite=yes as seen from my sip.conf: [mySIP] type=peer auth=md5 username=<UID> fromuser=<UID> fromdomain=&l...

...ting:15] > Queue("SIP/incoming-00000246", "myqueue1,,,,300,,,") in new stack > [Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class > 'default', on channel 'SIP/incoming-00000246' > > [Nov 21 08:18:26] pbx.c: Executing [mysip692 at CallFromQueue:1] > NoOp("Local/mysip692 at CallFromQueue-0000003c;2", "") in new stack > [Nov 21 08:18:26] app_queue.c: Called Local/mysip692 at CallFromQueue > [Nov 21 08:18:26] pbx.c: Executing [mysip692 at CallFromQueue:3] > Dial("Local/...

...it the quality is REALLY BAD. The obvious difference is that using directly the SJPhone i am using STUN, while when i am using Asterisk to connect to my SIP provider and the SJPhone to connect to Asterisk i have the following configuration for Asterisk. register => user:pass@sip.provider.com [mysip] host=sip.provider.com type=peer qualify=yes username=user secret=pass nat=yes disallow=all allow=ulaw I am using Asterisk 1.2.3. I think that i am missing something or misconfigure something because for sure its not matter of the ADSL since in both tests i am doing i am using the same circuit....

...l("SIP/20-0889c4d8", "SIP/mediant/1") in new stack -- Called mediant/1 -- SIP/mediant-088a1a18 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Congestion("SIP/20-0889c4d8", "") in new stack == Spawn extension (mysip, 111, 2) exited non-zero on 'SIP/20-0889c4d8' -- Executing Dial("SIP/24-0889c4d8", "SIP/mediant/0") in new stack -- Called mediant/0 my extension.conf file is exten => 43,1,Answer exten => 43,2,Dial(SIP/43) exten => 43,3,Hangup exten => 777,1,Answe...

...rst MSN msn=01234123457 ;incomingmsn=* incomingmsn=123457 isdnmode=msn controller=1 softdtmf=1 accountcode= context=isdn-in echosquelch=1 echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 {repeated for next 7 MSNs} And in extensions.conf I have: [globals] ISDN1=CAPI/123456 [mysip] ;GET OUTSIDE LINE (ISDN1 - dial 9) ignorepat => 9 exten => exten => _9.,1,Dial(${ISDN1}/${EXTEN:1}/b) exten => _9.,2,Playback(busy) exten => _9.,3,Hangup ***** I've tried using ISDN1=CAPI/contr1 but it makes no difference. I've tried leaving out the isdnmode=msn but it ma...

Hello list, Im writing my dial plan, in witch every SIP phone begins with 74 and has more 3 numbers (like 74XXX). So, I want to route all 74XXX calls to my sip channel. For this I wrote this line: exten => s,1,Dial(SIP/74118@193.136.252.5,30,r) but this way all calls go to 74118@193.136.252.5 ..... Then I tried: exten => s,1,Dial(SIP/${EXTEN}@193.136.252.5,30,r) but this way, the

...57] pbx.c: Executing [queue at pbx-routing:15] Queue("SIP/incoming-00000246", "myqueue1,,,,300,,,") in new stack [Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/incoming-00000246' [Nov 21 08:18:26] pbx.c: Executing [mysip692 at CallFromQueue:1] NoOp("Local/mysip692 at CallFromQueue-0000003c;2", "") in new stack [Nov 21 08:18:26] app_queue.c: Called Local/mysip692 at CallFromQueue [Nov 21 08:18:26] pbx.c: Executing [mysip692 at CallFromQueue:3] Dial("Local/mysip692 at CallFromQueue-0000003c...

Hi all I was tryed to connect to mysip.ch scs_client by siemens that isn't works well. Does anyones knows to work H/W or S/W applictations in asterisk SIP videosupport? Regards mack_jpn

...that > using directly the SJPhone i am using STUN, while when i am using > Asterisk to connect to my SIP provider and the SJPhone to connect to > Asterisk i have the following configuration for Asterisk. > > > register => user:pass@sip.provider.com > > [mysip] > host=sip.provider.com > type=peer > qualify=yes > username=user > secret=pass > nat=yes > disallow=all > allow=ulaw > > > I am using Asterisk 1.2.3. > > I think that i am missing something or misconfigure something becau...