Derek Beaumont
2003-Jul-08 08:10 UTC
[Asterisk-Users] Using multiple iconnecthere accounts
Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=>_91NXXNXXXXXX,1,StripMSD,1 exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the first account exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is the second account But that doesn't work. Has anybody tried this before?
Did asterisk register with both accounts ? "sip show registry" Can you post what happens on the console along with 'sip debug' ? Martin On Tue, 8 Jul 2003, Derek Beaumont wrote:> Has anybody out there tried to use two different iconnecthere accounts > with Asterisk? > What I want to do is use a second account if the first is busy. > I have tried the following: > > exten=>_91NXXNXXXXXX,1,StripMSD,1 > exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the > first account > exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is > the second account > > But that doesn't work. Has anybody tried this before? > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
Steven Critchfield
2003-Jul-08 08:29 UTC
[Asterisk-Users] Using multiple iconnecthere accounts
On Tue, 2003-07-08 at 10:10, Derek Beaumont wrote:> Has anybody out there tried to use two different iconnecthere accounts > with Asterisk? > What I want to do is use a second account if the first is busy. > I have tried the following: > > exten=>_91NXXNXXXXXX,1,StripMSD,1 > exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the > first account > exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is > the second account > > But that doesn't work. Has anybody tried this before?Isn't busy n+101 priority, or is it n+100? Basically you dial out similar to how you set up the busy portion of your voicemail. -- Steven Critchfield <critch@basesys.com>
Derek Beaumont
2003-Jul-08 09:51 UTC
[Asterisk-Users] Using multiple iconnecthere accounts
Asterisk has registered with both accounts: sip show registry Host Username Refresh State 213.137.73.178:5060 xxxxxxxx 120 Registered 213.137.73.178:5060 xxxxxxxx 120 Registered I can make one call just fine, but when I try to make the second call, I get an invalid extension error. When using the following configuration:>exten=>_91NXXNXXXXXX,1,StripMSD,1 >exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect >exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2I get the following output Executing Dial("Zap/4-1", "SIP/BYEXTENSION@iconnect") in new stack -- Called xxxxxxxxxxx@iconnect -- SIP/iconnect-cd45 is making progress passing it to Zap/4-1>show channels >Peer Username Call ID Seq (Tx/Rx) Lag JitterFormat>213.137.73.178 xxxxxxxxxx 6631b1e766b 00103/00000 00000ms 0000ms4>1 active SIP channel(s)This appears when I make the first call. I notice that I have a 0ms Jitter buffer. I am now curious as to how I create a jitter buffer in sip.conf? I have the following in the [general] section of sip.conf>jitterbuffer=yes >dropcount=3 >maxjitterbuffer=2500 >maxexccessbuffer=100Below is the output when I tried to call a second long distance number -- Executing Dial("Zap/4-2", "SIP/BYEXTENSION@iconnect") in new stack -- Called xxxxxxxxxxx@iconnect -- Got SIP response 480 "Temporarily not available" back from 213.137.73.178 -- SIP/iconnect-fde9 is circuit-busy == Everyone is busy at this time -- Executing Dial("Zap/4-2", "SIP/BYEXTENSION@iconnect2") in new stack -- Called xxxxxxxxxxx@iconnect2 sip show channels Peer Username Call ID Seq (Tx/Rx) Lag Jitter Format 213.137.37.178 xxxxxxxxxx 7047ee1a76b 00102/00000 00000ms 0000ms 2 213.137.73.176 xxxxxxxxxx 7b782a7b3dd 00103/00000 00000ms 0000ms 4 2 active SIP channel(s) *CLI> WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum retries exceeded on call 7047ee1a76b10c2e56afddfe6bc01895@xxx.xxx.xxx.xxx for seqno 102 (Request) == No one is available to answer at this time -- Sent into invalid extension 'xxxxxxxxxxx' in context 'outgoing' on Zap/4-2 -- Executing Playback("Zap/4-2", "TelError") in new stack -- Playing 'TelError' WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum retries exceeded on call 7047ee1a76b10c2e56afddfe6bc01895@67.70.231.220 for seqno 102 (Request) Any help is appreciated. Thank you for your time. ========OLD MESSAGE==========================>>Did asterisk register with both accounts ? >>"sip show registry" >> >>Can you post what happens on the console along with 'sip debug' ? >> >>MartinOn Tue, 8 Jul 2003, Derek Beaumont wrote:> Has anybody out there tried to use two different iconnecthere accounts > with Asterisk? > What I want to do is use a second account if the first is busy. > I have tried the following: > > exten=>_91NXXNXXXXXX,1,StripMSD,1 > exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the > first account > exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is > the second account > > But that doesn't work. Has anybody tried this before? > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
Derek Beaumont
2003-Jul-08 10:46 UTC
[Asterisk-Users] Using multiple iconnecthere accounts
>>Derek, tray this - it's working 100% with iconnect: >> >>exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@iconnect) >>exten => _91NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@iconnect2) >> >> >>Best regards >>LuboI tried that exact configuration, and now I can't even make a single call. What does EXTEN:1 do? Why is StripMSD not used?
Derek Beaumont
2003-Jul-08 12:31 UTC
[Asterisk-Users] Using multiple iconnecthere accounts
First off, sorry for using a mail client without the "in-reply-to" function. Second: I still can't make two calls using iconnecthere at the same time. Here is what I have tried: Attempt 1:>>exten=>_91NXXNXXXXXX,1,Dial,StripMSD >>exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect >>exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2Attempt 2:>>exten=>_91NXXNXXXXXX,1,Dial,StripMSD >>exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect >>exten=>_1NXXNXXXXXX,103,Dial,SIP/BYEXTENSION@iconnect2Attempt 3:>>exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@iconnect) >>exten => _91NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@iconnect2)Attempt 4: exten => _91NXXNXXXXXX,1,Dial,SIP/${EXTEN:1}@iconnect&SIP/${EXTEN:1}@iconnect2 So far nothing has worked. Another question I have is about jitter buffer. Is there a way to create a Jitter buffer in sip.conf? When I type sip show channels I get the following output: sip show channels Peer Username Call ID Seq (Tx/Rx) Lag Jitter Format 213.137.73.176 xxxxxxxxxx 5752cb7a55f 00103/00000 00000ms 0000ms 4 There is a section for Jitter, so I would imagine that there is some way to do it. Thank you for your time. Also, if anybody could suggest a good mail client for windows that is able to use the in-reply-to function, it would be helpful.