Re: [asterisk-users] audio glitches in conference
It looks to me that u are having clock synchronism problems due to the fact you are using Virtual Machine so u don't have an ISDN card generating clock. Are u using what was called ztdummie as clock source? Can't precise the name of it in chan_dahdi but u have it. What u report isn't new and is well known due to the fact u don't have a precise clock source for meetme.. You need to have chan_dahdi dummie... Hope it helps. Marco Mouta Enviada do dispositivo sem fios BlackBerry® -Original Message- From: Jeff Brower jbro...@signalogic.com Date: Wed, 24 Feb 2010 18:25:07 To: Jonathan Addlemanj...@redowl.ca Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] audio glitches in conference Jonathan- I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording (via eagi) after the conference has been established for half an hour or so, the stream received by the eagi script delayed by about 10 seconds. How did you measure the gaps? Using signal or speech analysis software to display the recording? If you measure number of samples between the gaps, does it correspond to multiples of RTP packet payload length (for example, for 8 kHz G711 multiples of 80 samples between gaps) ? -Jeff First, the preliminaries: I'm on a debian lenny system, using the 1:1.4.21.2~dfsg-3 asterisk package. This is a dedicated server - was running xen, but I've shut down all the domU's to test if they were interfering, so now there's no sharing going on. I've been testing with a simple eagi script to grab the audio from the conference: #!/bin/sh cat /dev/fd/3 /tmp/audio.raw I've been testing it using the following dialplan extensions: [test] exten = testeagi,1,Answer exten = testeagi,n,Wait(3) exten = testeagi,n,EAGI(testeagi.sh) exten = testmeet,1,Answer exten = testmeet,n,MeetMe(testconf,1qd) exten = testsound,1,Answer exten = testsound,n,Playback(testbeep-asterisk) (testbeep is just 30 seconds of sine wave) I've been trying things like this: originate Local/testso...@test extension teste...@test The recorded audio plays back fine - no glitches. (an example is at http://www.vecotourism.org/audio17.wav) originate Local/teste...@test extension testm...@test originate Local/testso...@test extension testm...@test This does have the glitches. (an example is at http://www.vecotourism.org/audio18.wav) What could be causing this? And is there anything else I could be doing to debug it? Thanks. -- Jon-o Addleman - http://www.redowl.ca -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio glitches in conference
Thanks to Tzafrir for the above mentiong documentation. FYI http://docs.tzafrir.org.il/dahdi-linux/README.html A PBX system should generally have a single clock. If you are connected to a telephony provider via a digital interface (e.g: E1, T1) you should also typically use the provider's clock (as you get through the interface). Hence one important job of Asterisk is to provide timing to the PBX. DAHDI ticks once per millisecond (1000 times per second). On each tick every active DAHDI channel reads and 8 bytes of data. Asterisk also uses this for timing, through a DAHDI pseudo channel it opens. However, not all PBX systems are connected to a telephony provider via a T1 or similar connection. With an analog connection you are not synced to the other party. And some systems don't have DAHDI hardware at all. Even a digital card may be used for other uses or is simply not connected to a provider. DAHDI cards are also capable of providing timing from a clock on card. Cheap x100P clone cards are sometimes used for that purpose. If all the above fail, you can use the module dahdi_dummy to provide timing alone without needing any DAHDI hardware. It will work with most systems and kernels. You can check the DAHDI timing source with dahdi_test, which is a small utility that is included with DAHDI. It runs in cycles. In each such cycle it tries to read 8192 bytes, and sees how long it takes. If DAHDI is not loaded or you don't have the device files, it will fail immediately. If you lack a timing device it will hang forever in the first cycle. Otherwise it will just give you in each cycle the percent of how close it was. Also try running it with the option -v for a verbose output. To check the clock source that is built into dahdi_dummy, you can either look at title of its span in /proc/dahdi file for a source: in the description. Or even run: strings dahdi.ko | grep source: -- Marco Mouta On Thu, Feb 25, 2010 at 8:15 AM, marco.mo...@gmail.com wrote: It looks to me that u are having clock synchronism problems due to the fact you are using Virtual Machine so u don't have an ISDN card generating clock. Are u using what was called ztdummie as clock source? Can't precise the name of it in chan_dahdi but u have it. What u report isn't new and is well known due to the fact u don't have a precise clock source for meetme.. You need to have chan_dahdi dummie... Hope it helps. Marco Mouta Enviada do dispositivo sem fios BlackBerry® -Original Message- From: Jeff Brower jbro...@signalogic.com Date: Wed, 24 Feb 2010 18:25:07 To: Jonathan Addlemanj...@redowl.ca Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] audio glitches in conference Jonathan- I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording (via eagi) after the conference has been established for half an hour or so, the stream received by the eagi script delayed by about 10 seconds. How did you measure the gaps? Using signal or speech analysis software to display the recording? If you measure number of samples between the gaps, does it correspond to multiples of RTP packet payload length (for example, for 8 kHz G711 multiples of 80 samples between gaps) ? -Jeff First, the preliminaries: I'm on a debian lenny system, using the 1:1.4.21.2~dfsg-3 asterisk package. This is a dedicated server - was running xen, but I've shut down all the domU's to test if they were interfering, so now there's no sharing going on. I've been testing with a simple eagi script to grab the audio from the conference: #!/bin/sh cat /dev/fd/3 /tmp/audio.raw I've been testing it using the following dialplan extensions: [test] exten = testeagi,1,Answer exten = testeagi,n,Wait(3) exten = testeagi,n,EAGI(testeagi.sh) exten = testmeet,1,Answer exten = testmeet,n,MeetMe(testconf,1qd) exten = testsound,1,Answer exten = testsound,n,Playback(testbeep-asterisk) (testbeep is just 30 seconds of sine wave) I've been trying things like this: originate Local/testso...@test extension teste...@test The recorded audio plays back fine - no glitches. (an example is at http://www.vecotourism.org/audio17.wav) originate Local/teste...@test extension testm...@test originate Local/testso...@test extension testm...@test This does have the glitches. (an example is at http://www.vecotourism.org/audio18.wav) What could be causing this? And is there anything else I could be doing to debug it? Thanks. -- Jon-o Addleman - http://www.redowl.ca
[asterisk-users] lawnmower man attack sip tag=Zerogij34 some one else notice this in 20th september or recently?
Dear all, According to: http://www.honeynor.no/2009/09/20/citibank-uk-number-was-target-for-a-lawnmower-telephone-attack-today/ Citibankhas been under a telephone calling attack in 20th september. Does anyone in asterisk community got any CDRs or logging of similar attacks as the one above mentioned ? Any one with logging of it or future information about the case ? Identified more detaills in this attack ? Citibank is or has been under a telephone calling attack latest 12 hours. Here I will explain the attack and how it was done. Have you seen the movie “lawnmower man”, when in the end, all phones rings in the who city? This was the aim for todays attack on Citibank in UK. The attack was simple, but probably effective when it was active. Send SIP INVITE to open SIP gateways and PBXs, who then will actually use the traditional phonesystem (POTS) to call the target. Suddenly you need DoS protection on your traditional POTS lines…. The SIP INVITE looks like this. INVITE sip:00442075005...@x SIP/2.0 Via: SIP/2.0/UDP 217.23.7.47:58585;branch=z9hG4bKaergjerugroijrgrg To: sip:x From: sip:217.23.7.47:58585;tag=Zerogij34 Call-ID: 213948958-34384780214-384...@217.23.7.47 CSeq: 1 INVITE Max-Forwards: 69 Contact: sip:s...@217.23.7.47:58585;transport=udp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE Content-Type: application/sdp Content-Length: 520 Session-Expires: 3600; Allow-Events: refer.. v=0 o=sip 2147483647 1 IN IP4 1.1.1.1 s=sip c=IN IP4 1.1.1.1 t=0 0 m=audio 29784 RTP/AVP 8 0 4 18 18 18 18 96 3 98 a=rtpmap:96 telephone-event/8000 a=sendrecva=ptime:20 a=rtpmap:18 G729AB/8000 a=rtpmap:18 G729B/8000 a=rtpmap:18 G729A/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723 Lets walk through the SIP packet and see what info we can get from it: A quick google search on the tag: Zerogij34 reveals that this attack has been around since at least 6th of August. The IP (217.23.7.47)from this packet should be located in Portugal but the other attacks originate from both UK and Netherlands. There is no User-Agent listed, so the packet is very likely crafted from toosl like sipsak or sipp. The codec list seems real, but they use an obscure address (1.1.1.1) for the RTP. If they would use their own IP address, it could case a small DoS with RTP traffic for every successful call.)The port 29784 is within the range of Cisco units (26 000-32 000) The other INVITES reveals that the attacker is trying to figure the extension to get a dial-tone: * INVITE sip:00442075005...@67.170.104.216 SIP/2.0 * INVITE sip:011442075005...@67.170.104.216 SIP/2.0 * INVITE sip:0442075005...@67.170.104.216 SIP/2.0 * INVITE sip:442075005...@67.170.104.216 SIP/2.0 * INVITE sip:0011442075005...@67.170.104.216 SIP/2.0 * INVITE sip:900442075005...@67.170.104.216 SIP/2.0 * INVITE sip:9011442075005...@67.170.104.216 SIP/2.0 * INVITE sip:90442075005...@67.170.104.216 SIP/2.0 * INVITE sip:442075005...@67.170.104.216 SIP/2.0 * and several more… But is this a DoS attack on Citibank? I doubt it. Why call the Citibank on a Sunday 5 a.m.? This is more likely that Citibank has lots of lines and therefore the SIP INVITES does not generate an error (busy or others). The attacker does not hear any ringtone, but he/she should see the 180 Ringing / 180 Session in Progress. Then he or she knows that he could actually get through to the PSTN on this SIP proxy. If it would be a ringing attack, why does the attacker just send one single SIP INVITE through each gateway that actually calls this destination? The machines with the attacking IP addresses should be put under surveillance to see who connects to these. They are probably just some bots in a larger network, but they need to relay back which gateways actually responded successfully. Sad to say, but I believe this is only the small beginning…. Looking forward to hearing from you guys ;) Cheers, -- Marco Mouta ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] lawnmower man attack ??
Dear all, According to: w w w .honeynor.no/2009/09/20/citibank-uk-number-was-target-for-a-lawnmower-telephone-attack-today/ Citibankhas been under a telephone calling attack in 20th september. Does anyone in asterisk community got any CDRs or logging of similar attacks as the one above mentioned ? Any one with logging of it or future information about the case ? Identified more detaills in this attack ? Citibank is or has been under a telephone calling attack latest 12 hours. Here I will explain the attack and how it was done. Have you seen the movie “lawnmower man”, when in the end, all phones rings in the who city? This was the aim for todays attack on Citibank in UK. The attack was simple, but probably effective when it was active. Send SIP INVITE to open SIP gateways and PBXs, who then will actually use the traditional phonesystem (POTS) to call the target. Suddenly you need DoS protection on your traditional POTS lines…. The SIP INVITE looks like this. INVITE sip:00442075005...@x SIP/2.0 Via: SIP/2.0/UDP 217.23.7.47:58585;branch=z9hG4bKaergjerugroijrgrg To: sip:x From: sip:217.23.7.47:58585;tag=Zerogij34 Call-ID: 213948958-34384780214-384...@217.23.7.47 CSeq: 1 INVITE Max-Forwards: 69 Contact: sip:s...@217.23.7.47:58585;transport=udp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE Content-Type: application/sdp Content-Length: 520 Session-Expires: 3600; Allow-Events: refer.. v=0 o=sip 2147483647 1 IN IP4 1.1.1.1 s=sip c=IN IP4 1.1.1.1 t=0 0 m=audio 29784 RTP/AVP 8 0 4 18 18 18 18 96 3 98 a=rtpmap:96 telephone-event/8000 a=sendrecva=ptime:20 a=rtpmap:18 G729AB/8000 a=rtpmap:18 G729B/8000 a=rtpmap:18 G729A/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723 Lets walk through the SIP packet and see what info we can get from it: A quick google search on the tag: Zerogij34 reveals that this attack has been around since at least 6th of August. The IP (217.23.7.47)from this packet should be located in Portugal but the other attacks originate from both UK and Netherlands. There is no User-Agent listed, so the packet is very likely crafted from toosl like sipsak or sipp. The codec list seems real, but they use an obscure address (1.1.1.1) for the RTP. If they would use their own IP address, it could case a small DoS with RTP traffic for every successful call.)The port 29784 is within the range of Cisco units (26 000-32 000) The other INVITES reveals that the attacker is trying to figure the extension to get a dial-tone: * INVITE sip:00442075005...@67.170.104.216 SIP/2.0 * INVITE sip:011442075005...@67.170.104.216 SIP/2.0 * INVITE sip:0442075005...@67.170.104.216 SIP/2.0 * INVITE sip:442075005...@67.170.104.216 SIP/2.0 * INVITE sip:0011442075005...@67.170.104.216 SIP/2.0 * INVITE sip:900442075005...@67.170.104.216 SIP/2.0 * INVITE sip:9011442075005...@67.170.104.216 SIP/2.0 * INVITE sip:90442075005...@67.170.104.216 SIP/2.0 * INVITE sip:442075005...@67.170.104.216 SIP/2.0 * and several more… But is this a DoS attack on Citibank? I doubt it. Why call the Citibank on a Sunday 5 a.m.? This is more likely that Citibank has lots of lines and therefore the SIP INVITES does not generate an error (busy or others). The attacker does not hear any ringtone, but he/she should see the 180 Ringing / 180 Session in Progress. Then he or she knows that he could actually get through to the PSTN on this SIP proxy. If it would be a ringing attack, why does the attacker just send one single SIP INVITE through each gateway that actually calls this destination? The machines with the attacking IP addresses should be put under surveillance to see who connects to these. They are probably just some bots in a larger network, but they need to relay back which gateways actually responded successfully. Sad to say, but I believe this is only the small beginning…. Looking forward to hearing from you guys ;) Cheers, -- Marco Mouta ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 428 Loop Detected
It's so uncommon for me fxs and fxo cards and based on the reference of sip.conf files and accounts i totally missed last paragraph where it was mentioned only analogue lines and fxs (phone). my appologies. E1 and BRIs and sip trunks have been overloading my last month of work. cheers, -- Marco Mouta 2009/3/16 Steve Totaro stot...@totarotechnologies.com: Again, if I am interpreting this correctly, he is not using SIP. A four port card 2fxo/2fxs means to me that he is not using SIP at all. If by card, you mean some kind of SIP gateway, then I misunderstood and the problem, but seeing DAHDI channels leads me to believe that SIP is not required and actually causing your problems. SIP is a protocol for VoIP, DAHDI/Zaptel is TDM (analog POTS in this case)... If you had a SIP device, it would be connected to the data network, not a phone line. Can you just plug your phone into a regular landline jack and get dialtone? If so, forget SIP for now. Comment out or delete all your sip.conf peers since you are not using SIP. Change your dialplan to not (Dial/SIP but (Dial/DAHDI/1,10) and the correct channel to your FXS port that the phone is connected to. Thanks, Steve Totaro On Sun, Mar 15, 2009 at 9:20 PM, Marco Mouta marco.mo...@gmail.com wrote: Hi, problem is that you are saying that phone in sip.conf is at the same ip address of your asterisk box so you are dialing into a loop to your self asterisk box [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 what you need is: [phone] type=friend context=phone1 secret=g00dpazzwerd dtmfmode=rfc2833 host=dynamic ;configuring your codecs (i don't know what else you have configured, just preventing audio for you) disallow=all allow=ulaw allow=alaw allow=gsm Dial sip/phone is enough too.. [from-pstn] ;include = default exten = s,1,Dial(SIP/phone,10) exten = s,2,Voicemail(line) exten = s,3,Hangup hope it helps. don't forget to asterisk reload on cli. Looking forward to hearing from you. cheers -- Marco Mouta On Sun, Mar 15, 2009 at 10:28 PM, Asif Iqbal vad...@gmail.com wrote: Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqb...@improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 [line] type=friend context=line1 secret=anothers33cret bindport=5061 host=192.168.1.106 dtmfmode=rfc2833 iqb...@improvise:/etc/asterisk$ cat extensions.conf [default] exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,Playback(tt-monkeys) exten = s,4,Hangup [from-internal] include = default [phone1] [from-pstn] ;include = default exten = s,1,Dial(SIP/ph...@phone,10) exten = s,2,Voicemail(line) exten = s,3,Hangup [line1] So my home land line is going to the FXO port and my home phone is hanging off of FXS port. Here are the contexts for my fxo/fxs card improvise*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudo default default 1 from-internal default 2 from-internal default 3 from-pstn default 4 from-pstn default I want to call from my cell and make my home phone ring and if I dont pickup in 10 secs I want the call go to my voicemail. But I am getting a loop detect. The debug output is attached. What am I doing wrong? -- Asif Iqbal PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 428 Loop Detected
Hi, problem is that you are saying that phone in sip.conf is at the same ip address of your asterisk box so you are dialing into a loop to your self asterisk box [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 what you need is: [phone] type=friend context=phone1 secret=g00dpazzwerd dtmfmode=rfc2833 host=dynamic ;configuring your codecs (i don't know what else you have configured, just preventing audio for you) disallow=all allow=ulaw allow=alaw allow=gsm Dial sip/phone is enough too.. [from-pstn] ;include = default exten = s,1,Dial(SIP/phone,10) exten = s,2,Voicemail(line) exten = s,3,Hangup hope it helps. don't forget to asterisk reload on cli. Looking forward to hearing from you. cheers -- Marco Mouta On Sun, Mar 15, 2009 at 10:28 PM, Asif Iqbal vad...@gmail.com wrote: Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqb...@improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 [line] type=friend context=line1 secret=anothers33cret bindport=5061 host=192.168.1.106 dtmfmode=rfc2833 iqb...@improvise:/etc/asterisk$ cat extensions.conf [default] exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,Playback(tt-monkeys) exten = s,4,Hangup [from-internal] include = default [phone1] [from-pstn] ;include = default exten = s,1,Dial(SIP/ph...@phone,10) exten = s,2,Voicemail(line) exten = s,3,Hangup [line1] So my home land line is going to the FXO port and my home phone is hanging off of FXS port. Here are the contexts for my fxo/fxs card improvise*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudo default default 1 from-internal default 2 from-internal default 3 from-pstn default 4 from-pstn default I want to call from my cell and make my home phone ring and if I dont pickup in 10 secs I want the call go to my voicemail. But I am getting a loop detect. The debug output is attached. What am I doing wrong? -- Asif Iqbal PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?
try to set in your zapata.conf
overlapdial=yes
then in your asterisk cli
reload chan_zap.so
--
Marco Mouta
On Fri, Feb 13, 2009 at 9:21 AM, joek...@gmail.com wrote:
Default FreePBX context,
[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did
include = ext-did-post-custom
include = from-did-direct; MODIFICATOIN (PL) for findmefollow if
enabled, should be bofore ext-local
include = ext-did-catchall; THIS MUST COME AFTER ext-did
exten = fax,1,Goto(ext-fax,in_fax,1)
The call seems to be going here
[ext-did-catchall]
include = ext-did-catchall-custom
exten = s,1,Noop(No DID or CID Match)
exten = s,n(a2),Answer
exten = s,n,Wait(2)
exten = s,n,Playback(ss-noservice)
exten = s,n,SayAlpha(${FROM_DID})
exten = s,n,Hangup
exten = _.,1,Set(__FROM_DID=${EXTEN})
exten = _.,n,Noop(Received an unknown call with DID set to ${EXTEN})
exten = _.,n,Goto(s,a2)
exten = h,1,Hangup
; end of [ext-did-catchall]
--
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Re: [asterisk-users] Pressing 0 to get an external line
Hello, please read bellow: On Tue, Sep 9, 2008 at 11:04 PM, Christian Victor [EMAIL PROTECTED] wrote: Hi Asterisk users! I have a little problem with an Asterisk 1.4.22 installation for a customer. The PBX is connected to an E1 line and we have a few snom 300 attached to it. The goal is to emulate traditional german PBX behaviour wich is the play a stuttered internal dialtone after pickup and accept all internal extensions. If a specific get office digit - usually 0 - is pressed you get the normal dialtone. Now you can enter the external telephone number end get connected immediately without pressing dial on the snom phone. My problem here is first to implement this internal/external dialtone You can use DISA application without authentication to fake the dialtone of traditional german PBX. and second to make the phone dial without pressing dial on the phone. This might be a first scratch exten = _X.,1,Answer exten = _X.,2,DigitTimeout(5) exten = _X.,3,ResponseTimeout(10) exten = _X.,4,DISA(no-password|mycontext) And with this DigitTimeout and Response Timeout you may solve the need to press dial I guess... Looking forward to hearing from you. Kind regards, Mouta Because the length of telephone numbers in germany is not constant we will have to implement this with a timeout or by somehow overlap-dialing digit by tigit to the E1. I hope one of you already implemented this behaviour or can maybe push me in the right direction. Thanks a lot Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Next step in extensions.conf after answer the phone in Queue
May be I'm wrong but:*
timeout - the maximum time, in seconds, the call will wait in the queue.
When this time expires, the next extension, by priority, will be executed.
By default the timeout is set to 300 seconds.
So you clearly have two ways to feed your database with your statistics:
If (agent is busy)
{
You will get timeout and you will go throuhg the next priority of your
dialplan
}
else
{
You can use the exten= h to insert your database info correctly just after
your customer hangup the call
}
Also notice that
This application sets the following channel variable upon completion:
QUEUESTATUSThe status of the call as a text string, one of TIMEOUT |
FULL | JOINEMPTY | LEAVEEMPTY | JOINUNAVAIL | LEAVEUNAVAI
run on your asterisk CLI show application Queue
I hope it helps.
Kind regards,
Mouta
*
On Wed, Apr 23, 2008 at 6:39 PM, AnDY [EMAIL PROTECTED] wrote:
I want to log in database some info ( total agents logged in, busy
agents, time ... ). I have some variables and checking them.
Let me explain it from beginning:
Somebody call the queue and everyone is busy, i need to play to caller
that everyone is busy and he should call later, and log this situation.
So I have to variables. Total agents available (TQC) to pick up and busy
agents (BQC). If everyone is busy than this to variables matches and the
caller has to wait. This I need to log with this two numbers.
If BQC != TQC I need just log the numbers nothing else.
Hopefully it is clear to you.
Maybe there is another solution how to do that.
Btw. I am putting this stats in MySQL database.
Andy
Al Baker napsal(a):
Why would you want a channel to continue after the caller has hung up.
I clearly am missing something here because I can't see what good that
would be. What do people do with this Continued Channel ?
What is is used for ? How Does having it help you ? ???
Atis Lezdins wrote:
Queue will continue if called person hangs up (and there's no option).
If caller hangs up, call goes to h extension in same context. Just the
same way as Dial with 'g'. There's a change in 1.6 that allows called
channel to continue if caller hangs up, so probably something like
this could be applied also to Queue (or was that actually working with
using Local channels?).
Regards,
Atis
On Wed, Apr 23, 2008 at 7:13 PM, AnDY [EMAIL PROTECTED] wrote:
Thank you for your answer.
But the Dial command has a option 'g' which means that after succes
will
proceed next priorities in the dialplan. Is there something also for
Queue() because according to manual there is no option for it. So I
am
looking for some other solution.
Andy
Tony Mountifield napsal(a):
In article
[EMAIL PROTECTED],
[EMAIL PROTECTED] wrote:
Hello everybody.
I was looking for the solution but nothing found. I have this in
my
extensions.conf:
exten = 233,1,SetAccount(queue1)
exten = 233,2,Queue(queue1|rn)
exten = 233,3,NoOp(${QUEUESTATUS})
exten = 233,4,NoOp(${DIALSTATUS})
But when the call is placed in the queue and somebody answer it,
it will
throw an error:
== Spawn extension (default, 211, 4) exited non-zero on
'Local/[EMAIL PROTECTED],2'
And no other command in extensions is executed.
Any suggestions?
Queue() is like Dial(), in that if it succeeds in connecting to
someone,
it will not return to the next priority in the dialplan. However,
if you
define an 'h' extension, that will get executed when the call is
complete:
exten = 233,1,SetAccount(queue1)
exten = 233,2,Queue(queue1|rn)
exten = 233,3,NoOp(${QUEUESTATUS})
exten = 233,4,NoOp(${DIALSTATUS})
exten = h,1,NoOp(${QUEUESTATUS})
exten = h,2,NoOp(${DIALSTATUS})
Cheers
Tony
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Re: [asterisk-users] call screening feature
Your solution is Asterisk Manager Interface http://www.voip-info.org/wiki-Asterisk+manager+API On Tue, Mar 18, 2008 at 6:24 AM, Janu Mukherjee [EMAIL PROTECTED] wrote: Hi, I have our software with SIP running on it.I configured asterisk server as proxy. How do I implement the call screening features(incoming and outgoing) using asterisk server.Please suggest me how to proceed on this. Thanks Regards, Jahnavi. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] php web chat + asterisk - callcenter
I would recommend you Asterisk for Voice and Video and XMPP for Chat. Asterisk in parallel with Jabberd2 (XMPP server) may feet your requirements, and if you use a XMPP MSN Transport Gateway you can do even more. On Mon, Mar 17, 2008 at 5:50 PM, Carlos Carvalhar [EMAIL PROTECTED] wrote: Hello, How can I make a live chat (mainly text, but with voice/video chat if possible) interacting with asterisk? Can asterisk control simultaneously the queue between people calling by phone and people by web chat? At my work, there is a call center using asterisk to control the queue of the clients (by phone) already. This part is ok. But now I need to make a chat room at the website and someone of the call center will need to answer that client. So my doubt is how to implement a solution that identifies an operator who is free and put him to talk by chat and then make him busy to phone calls. After the web chat is finished, the operator turns automatically free again. I'm planning to use php to set an asterisk variable telling the agent is free or busy. Can you tell me the asterisk apis involved with busy agents? Eg.: how do I set one agent as busy? I can set it by php, don't I? Is there any software like this one, Centriphone Millennium, for free? http://www.vocalcom.com/asterisk.html Is there any free solution? Where can I find information about how to settle asterisk variables (to get and to set) with php programming? I need to make a php page that settles a property of asterisk in runtime. Is it possible? How do I do it? Thanks in advance, Carlos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium certified asterisk professional linkedin group
Dear all, I've created a digium certified asterisk professional - dCAP linkedin group for anyone, dCAP, interested: http://www.linkedin.com/e/gis/60298/39AE1350DBF3 Best regards, Marco Mouta dCAP November 2006 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: VoIP SLA for SIP trunking - SMEs
Hi guys, I know that this could be considered a bit off the topic, I've just posted this topic at VoIPSEC mailing list but I just thought this could be very interesting for Asterisk community members so I'm posting it here too. So the point is for traditional telephony we expect service availability of 99,999% and what about VoIP providers around the world what is currently available? I know that some operators are not owners of the infrastructure so this becomes even harder to warranty the high level requirements that telephony demands. The best info and explanation I've found until now is: from bandwidth at http://www.bandwidth.com/pdf/voip/bandwidth_voip_sla_062105.pdf It would be interesting to discuss the recommend values needed and available at SLA agreements: Max Latency Max Jitter Max Packet Lost Mean Time Between Failures Mean Time To Repair Mean Opinion Score at least 4 ? Service Availability, this point as you probably are aware is very important, what I just notice a few time a go is that the difference between 99,9% and 99,999% is big! 99,9%- Max time for Outage during one month is 43,2 minutes , considering 30 days per month 99,999% - Max time for Outage during one month is 0,432 minutes If any of you around the world is aware of this values for VoIP SLAs I would be thankful to exchange and discuss this info. Thanks in advance. Best regards, Marco Mouta -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording on Hanup
What makes you decide wehter or not you want to keep the recorded file? Is the fact that the user hangup the call before the 30 seconds or the fact that he really talked? As far as I know current version of asterisk doesn't allow you to detect who end up the call. Of course you may use some tricks for these, I mean you may set the IVR to record a bit more than 30 seconds, then when the call hangs up when you reach the h extension in you diaplan you may check the answered time of the call. If your call has an answered time duration lower than 30 seconds, for sure was the caller who hangup the call. Hope it helps. On Dec 18, 2007 8:04 PM, Jamshed Zaidi [EMAIL PROTECTED] wrote: yes, the senario is this when user gets a call IVR starts playing and after hearing beep user starts recording message for 30 seconds(call duration is for 30 seconds). What i want is During 30 seconds if user does hangup his/her call then message should be recorded otherwise(after timeout) message is discarded. Is there any thing that will help me...??? currently I am doing the same thing on pressing 1 with php agi script and its working fine. On 12/19/07, Marco Mouta [EMAIL PROTECTED] wrote: What do you mean with record a call on hangup? If the calling party ends the call you want to keep recorded file? On Dec 18, 2007 6:27 PM, Jamshed Zaidi [EMAIL PROTECTED] wrote: Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps me in resolving this problem ??? -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All trunk are busy please try your call again later
Post: Asterisk CLI : sip show peers Asterisk CLI : zap show channels Asterisk CLI: zap show status As well as your extensions.conf Are you able to ping you GSM gateway? is connected via SIP or Telephony interface card? Best regards, Mouta On Dec 18, 2007 10:47 AM, Lolu Gbenga [EMAIL PROTECTED] wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox Phones Home
In http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home is said Kerry Garrison that: Both trixbox and FreePBX have phone-home mechanisms in them. So does FreePBX phones home too? On Dec 17, 2007 4:27 AM, Than Taro [EMAIL PROTECTED] wrote: As I pointed out here last night, there is also a very serious security vulnerability associated with this. Example: An attacker could compromise the script that is used on the remote host, and set it to force clients that connect to run a command such as rm -rf /. There are about half a dozen ways I could see this being abused - in either a one off or an every installation scenario. Fonality has yet to acknowledge this aspect of the issue - and I fear that they never will. See: http://voipsa.org/pipermail/voipsec_voipsa.org/2007-December/002522.html P.S.: On behalf of Rob (of FreePBX fame), I'd like to also point out this this is something that was added to trixbox, and not FreePBX. Quoting Rob: when someone mistakenly says 'trixbox does...' they usually mean 'freepbx does...' as FreePBX is the GUI Trixbox uses to configure Asterisk. In this instance, that is not the case - it is only a trixbox issue. From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com; [EMAIL PROTECTED] Date: Sun, 16 Dec 2007 20:53:53 -0500 Subject: [asterisk-users] Trixbox Phones Home I just read on Slashdot (at http://yro.slashdot.org/article.pl?sid=07/12/16/43 ) that Trixbox has been phoning home with statistics about their installations, as a Trixbox user exposed in Trixbox Phones Home at http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- The best games are on Xbox 360. Click here for a special offer on an Xbox 360 Console. Get it now! http://www.xbox.com/en-US/hardware/wheretobuy/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox Phones Home
Thanks Tzafrir! I really appreciate Free PBX. Keep on going your good job. Best regards, Mouta On Dec 18, 2007 11:59 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Dec 18, 2007 at 11:38:03AM +, Marco Mouta wrote: In http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home is said Kerry Garrison that: Both trixbox and FreePBX have phone-home mechanisms in them. So does FreePBX phones home too? And if you read further down that thread you would have seen the reply by philippel of FreePBX: ... | The only time this happens is when an online update is initiated by you, | or if you have chosen to receive update notifications since those are | nothing more then a cron Job that does exactly what Check for Online | Updates does in the GUI. ... -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording on Hanup
What do you mean with record a call on hangup? If the calling party ends the call you want to keep recorded file? On Dec 18, 2007 6:27 PM, Jamshed Zaidi [EMAIL PROTECTED] wrote: Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps me in resolving this problem ??? -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load Balancing over 2 E1 Lines
Why not Random application available in Asterisk ? quite simple I believe. asterisk1*CLI show application Random -= Info about application 'Random' =- [Synopsis] Conditionally branches, based upon a probability [Description] Random([probability]:[[context|]extension|]priority) probability := INTEGER in the range 1 to 100 best regards, Marco Mouta On Dec 12, 2007 8:08 AM, Eric Delaporte [EMAIL PROTECTED] wrote: Hi @ all, i set a server to a costumer of mine with a TE207P for use with 2 E1 Lines. I set them together into one group in zaptel/zapata.conf The point is now, the customer has a free-volumina of 60k minutes per month, per line. How can i make a kind of load balancing, that both lines will be trafficed the same way ? I read something about DIAL(Zap/r1/…) for using round robin, and it seems to work. Is there any other possible way to make sure that all lines are used in the same amount of minutes? Thanks in regard, Eric ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rollback procedure requirements before asterisk upgrade
Dear all, I've a live system that needs to be upgraded but, before I proceed to the upgrade I want to assure the rollback process. That's why I'm requesting your feedback, in fact this asterisk in live system isn't going so bad but the upgrade is essential NOTICE that the upgrade will keep the same version 1.2 not from 1.2 to 1.4 Requirements: -backup /usr/sbin/asterisk -backup /usr/lib/asterisk/modules Download asterisk 1.2.25 and compile it. This way I've tested and seems to work fine in my Virtual Machine lab. The only issue i found was one module that is loaded with my modules.confthat I needed to copy from the backup /usr/lib/asterisk/modules and give the right permissions. Am I missing something? best regards, Marco Mouta -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CAPI didn't get a frame | avoiding initial deadlock | multiple instances of Asterisk
Hi guys, First of all, I know that this server must be upgraded asap, I'm just wondering if anyone of you has already faced this problem and , if so, would the upgrade solve my problems... CAPI version 0.6 Asterisk 1.2.5 AGI scripts are being used Main problems: -Dropped Calls - ps aux | grep asterisk shows that asterisk (that is started with safe_asterisk) is generating multiple instances of asterisk by it's self. This may be caused by AGI scripts http://bugs.digium.com/view.php?id=8086? Extracted from /var/log/asterisk/full : Dec 6 10:11:07 DEBUG[5858] channel.c: Didn't get a frame from channel: CAPI/ISDN1/141-721 Dec 6 10:11:07 DEBUG[5858] channel.c: Bridge stops bridging channels CAPI/ISDN1/141-721 and SIP/741411-57bf Dec 6 10:11:07 DEBUG[5858] chan_sip.c: update_call_counter(741411) - decrement call limit counter Dec 6 10:11:07 VERBOSE[5858] logger.c: == Spawn extension (from-trunk, 141, 111) exited non-zero on 'CAPI/ISDN1/141-721' Dec 6 10:11:07 VERBOSE[5858] logger.c: -- Executing NoOp(CAPI/ISDN1/141-721, from-trunk - h - 112 - 112) in new stack Dec 6 10:11:07 VERBOSE[5858] logger.c: -- Executing AGI(CAPI/ISDN1/141-721, set_callerid.agi) in new stack Dec 6 10:11:08 DEBUG[5858] res_agi.c: CAPI/ISDN1/141-721 hungup Dec 6 10:11:08 VERBOSE[5858] logger.c: == Spawn extension (from-trunk, h, 2) exited non-zero on 'CAPI/ISDN1/141-721' Dec 6 10:11:08 DEBUG[2692] channel.c: Avoiding initial deadlock for 'CAPI/ISDN1/141-721' Dec 6 10:11:08 DEBUG[2692] channel.c: Avoiding initial deadlock for 'CAPI/ISDN1/141-721' Dec 6 10:13:02 DEBUG[5878] channel.c: Didn't get a frame from channel: SIP/741411-f7c1 Dec 6 10:13:02 DEBUG[5878] channel.c: Bridge stops bridging channels CAPI/ISDN1/141-722 and SIP/741411-f7c1 Dec 6 10:14:15 DEBUG[5902] channel.c: Didn't get a frame from channel: CAPI/ISDN1/141-723 Dec 6 10:14:15 DEBUG[5902] channel.c: Bridge stops bridging channels CAPI/ISDN1/141-723 and SIP/741411-4d0f Dec 6 10:14:34 DEBUG[2692] channel.c: Avoiding initial deadlock for 'CAPI/ISDN1/141-725' Dec 6 10:14:34 DEBUG[2692] channel.c: Avoiding initial deadlock for 'CAPI/ISDN1/141-725' Dec 6 10:14:54 VERBOSE[5941] logger.c: -- SIP/741411-26c2 answered CAPI/ISDN1/141-726 Dec 6 10:15:15 DEBUG[5941] channel.c: Didn't get a frame from channel: SIP/741411-26c2 Dec 6 10:15:15 DEBUG[5941] channel.c: Bridge stops bridging channels CAPI/ISDN1/141-726 and SIP/741411-26c2 Dec 6 10:45:29 DEBUG[6235] channel.c: Didn't get a frame from channel: SIP/741411-9981 Dec 6 10:45:29 DEBUG[6235] channel.c: Bridge stops bridging channels CAPI/ISDN1/132-72d and SIP/741411-9981 I definitely must go through CAPI 1.0 and asterisk 1.2.25, but would be interesting to learn from this problem before upgrading, that's why I'm posting on the list. Could it be possible that the multiple instances of Asterisk that are started due to the bug ( http://bugs.digium.com/view.php?id=8086, solved in asterisk 1.2.13) are causing all the troubles, making this multiple instances try to access same asterisk channel (leading us to Avoiding deadlock messages) ? I mean applying the patch might solve the problems instead off all system upgrade? Best regards, Marco Mouta -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk to connect 2 locations with legacy PBX
Yes Asterisk feets your needs 100%. You need 2 Asterisk each one with Telephony cards (ISDN or Analog) and I would recommend you to setup an IAX trunk to interconnect both servers. Each Asterisk will act as a Media Gateway. Look for IAX trunk over the wiki it's all there. good luck. Best regards, Marco Mouta On Dec 10, 2007 12:24 PM, Kovář Jan [EMAIL PROTECTED] wrote: Hello. I am going through the documentation and trying to find if asterisk can help me in my case. It is quite difficult to find answer because I do not know the exact question. I have two location. Each in different country. Both locations have Siemens HiPath - different type and software. I can not use card that would allow me to connect those PBXs using SIP. But I have some free ISDN and analog ports in both PBXs. Is it possible to use Asterisk on each side with some ISDN or analog card to connect to each PBX and enable users to call through the asterisk shortcut? The users would not dial +country code whole number, but for example 99 + internal number in second location to use Asterisk instead of conventional telecom line. Could this be done using Asterisk? What should I look for in documentation? Thank you Best regards Jan Kovář IT koordinátor __ Meggle, s.r.o., Gotthardská 4/37, CZ-160 00 Praha 6 tel +420 729 828 128 fax +420 224 312 184 gsm +420 724 112 882 [EMAIL PROTECTED] www.meggle.cz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange ISDN-problem with incoming calls out of the same city
Does this number (you are dialing) has been ported from a different Telco? When you dial from the other city and you get service not available you may be dialing from a different Telco that either has no route aggreement for the dialed network, or the number portability database (of Out of city Operator) is not up to date. Can you confirm this two things? On Dec 5, 2007 10:05 PM, Stefan Guenther [EMAIL PROTECTED] wrote: Hi, after I fixed my problem with the playback() application, I now have the next strange one. When I dial the number of our client, located in another town, I get a connection to the asterisk server, I can talk to my client or listen to his mailbox. If some in the town of this client calls him, he gets the ISDN error service not available. Out office is connected to he office via vpn, and so I connected a sip phone to his asterisk server. Now this phone is in the same town and when I dial his external number (to make sure it is an ISDN connection), I hear a single ring tone, then the phone is connected but I here nothing. Here is the output of capi debug: -- Executing [EMAIL PROTECTED]:1] Set(SIP/user1-0827eb08, CALLERID(num)=7253940397) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/user1-0827eb08, CAPI/g1/7253940397:940388|60|tr) in new stack -- Called g1/7253940397:940388 == ISDN5#01: Incoming call '072539403' - '940388' -- Executing [EMAIL PROTECTED]:1] Answer(CAPI/ISDN5#01/940388-3, ) in new stack == ISDN5#01: Answering for 940388 -- Executing [EMAIL PROTECTED]:2] Wait(CAPI/ISDN5#01/940388-3, 1) in new stack == ISDN5#01: Setting up echo canceller (PLCI=0x305, function=1, options=4, tail=64) == ISDN5#01: Setting up DTMF detector (PLCI=0x305, flag=1) -- ISDN5#01: Echo canceller successfully set up (PLCI=0x305) == ISDN5#02: Setting up echo canceller (PLCI=0x205, function=1, options=4, tail=64) == ISDN5#02: Setting up DTMF detector (PLCI=0x205, flag=1) -- CAPI/ISDN5#02/940388-2 answered SIP/user1-0827eb08 -- ISDN5#02: Echo canceller successfully set up (PLCI=0x205) -- Executing [EMAIL PROTECTED]:3] GotoIfTime(CAPI/ISDN5#01/940388-3, 17:00-18:00|*|*|*?from-extern|940388|6) in new stack -- Executing [EMAIL PROTECTED]:4] Dial(CAPI/ISDN5#01/940388-3, SIP/VERKAUF|20|tr) in new stack -- Called VERKAUF -- SIP/VERKAUF-082841a0 is ringing -- SIP/VERKAUF-082841a0 is ringing -- SIP/VERKAUF-082841a0 is ringing == ISDN5#02: CAPI Hangingup for PLCI=0x205 in state 2 == Spawn extension (local, 940388, 2) exited non-zero on 'SIP/user1-0827eb08' What is the difference between an isdn call starting in the same area and a call from my office? We are using a EICON DIVA Server 4 BRI with the current driver and an asterisk 1.4.13. Here is the start of the capi.conf [General] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 language=de immediate=yes faxdetect=off [ISDN1] incomingmsn=* context=from-extern ntmode=yes controller=1 group=1 callgroup=1 accountcode=ISDN1 echocancel=yes echosquelch=1 echotail=64 devices=2 Thanks for any suggestions, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium and Asterisk
I got one of this boards and I got it successfully replaced by Avanzada7 (Digium official reseller) immediately. On Nov 24, 2007 6:46 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Actually if you rule out all the clone tormenta cards (nothing wrong.. but very dated design... I wouldnt buy one today) the Digium cards aren't too expensive. Those tormenta cards are the ones you see for $300-400 typically. Some people like Digium others Sangoma. Personally I'm a Sangoma man. Some people report certain main boards and Dell servers aren't compatible with some digium cards. According to a post here on the mailing list someone from Digium implied that they will replace cards with these conflicts with newer model card that does not have these conflicts... your millage may vary I don't believe that forum posting was made in any official capacity but I also doubt that Digium would not do something to correct an issue for an item under warranty. On Nov 22, 2007 8:03 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; Is Digium the best telephony cards to be used with Asterisk? The prices are some how high, any suggestion? Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium and Asterisk
Digium Cards have been just great on my experience and their support has been simply the best one, via IAX (free Call) Remote Acess and hardware config review and troubleshooting. Many Thanks to Digium and their official reseller for Portugal and Spain Avanzada7 great work! Best regards, Marco Mouta ps. Do not forget that when you buy digium cards you are supporting the asterisk development. On Nov 22, 2007 1:03 PM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; Is Digium the best telephony cards to be used with Asterisk? The prices are some how high, any suggestion? Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] route INVITE sip:[EMAIL PROTECTED]
Hi,
I would suggest you to use Asterisk Application SIPGetHeader in your
Dialplan for incoming calls from plugandtel.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPGetHeader
Something like
*
exten=_[a-z].,1,SIPGetHeader(Var_TO=To)
exten=_[a-z].,2,Dial(SIP/${Var_TO})
Please be aware that this application as a different name in asterisk 1.4.
Learn from CLI show application ?
Hope it helps,
Best regards,
MoutaPT
*
On Nov 14, 2007 3:02 PM, Marc LEURENT [EMAIL PROTECTED] wrote:
We are using 2 different incoming trunks.
The first one is alsion.com and is sending INVITE with phone number in
the INVITE line whereas plugandtel put the callee number only inside the
To: Section.
Marco Mouta a écrit :
Could you describe in detail how did you fall into this situation, I
mean
the real example which SIP phone sends this invite? Is registered in
asterisk? it is a non-registered sip phone trying to dial a sip user at
your
* box?
If this is an issue with a specific hardware outside of your asterisk,
may
be something not well configured ... describe it a bit more in detail.
If you don't have anyworkaround for this Invite format I would use
OpenSER
in front of Asterisk to handle this invites and replace to SIP URI with
info
from the tag TO: ...
Any way if you provide more details may be someone in the Mailing list
is
able to help u out;)
Best regards
MoutaPT
On Nov 13, 2007 6:14 PM, Marc LEURENT [EMAIL PROTECTED] wrote:
Good evening!
I was wondering one thing,
I'm using freepbx to configure my asterisk server and I have a problem
with some inbound calls.
When I receive a call to an INVITE sip:[EMAIL PROTECTED] I an set an
inbound route! It matches a DID number.
How can I route an INVITE sip:[EMAIL PROTECTED] The number only appear in
the
To: Section.
Thanks!
Example:
With this one, I cannot route it (there is only the number to be
reached
in the To: section)
#
U 217.36.112.145:5060 - 192.168.95.235:5060
INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0.
Allow: UPDATE,REFER,INFO.
Call-ID: [EMAIL PROTECTED]
Contact: sip:217.66.118.145:5060.
Content-Type: application/sdp.
CSeq: 34878212 INVITE.
From: 0614740696
sip:[EMAIL PROTECTED]
;user=phone;tag=02975-US-0223ae6e-67d6c4495.
Max-Forwards: 31.
To: sip:[EMAIL PROTECTED];user=phone.
User-Agent: Cirpack/v4.41c (gw_sip).
Via: SIP/2.0/UDP 217.36.112.145:5060;branch=z9hG4bK-744D-33B812.
Content-Length: 303.
.
Whereas with this one I can do it! (there is a number in the INVITE)
#
U 87.98.202.114:5060 - 192.168.95.235:5060
INVITE sip:[EMAIL PROTECTED] SIP/2.0.
Via: SIP/2.0/UDP 87.98.202.114:5060;branch=z9hG4bK1fd2c6b4;rport.
From: 0158136741 sip:[EMAIL PROTECTED];tag=as25391ca7.
To: sip:[EMAIL PROTECTED].
Contact: sip:[EMAIL PROTECTED].
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Date: Tue, 13 Nov 2007 18:07:00 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Content-Type: application/sdp.
Content-Length: 233.
.
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--
Esta mensagem (incluindo quaisquer anexos) pode conter informação
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esta mensagem por engano, por favor informe o emissor e elimine-a
imediatamente. Obrigado.
This e-mail message is intended only for individual(s) to whom it is
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proprietary, or otherwise exempt from disclosure under applicable law. If
you believe you have received this message in error, please advise the
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Re: [asterisk-users] route INVITE sip:[EMAIL PROTECTED]
Could you describe in detail how did you fall into this situation, I mean the real example which SIP phone sends this invite? Is registered in asterisk? it is a non-registered sip phone trying to dial a sip user at your * box? If this is an issue with a specific hardware outside of your asterisk, may be something not well configured ... describe it a bit more in detail. If you don't have anyworkaround for this Invite format I would use OpenSER in front of Asterisk to handle this invites and replace to SIP URI with info from the tag TO: ... Any way if you provide more details may be someone in the Mailing list is able to help u out;) Best regards MoutaPT On Nov 13, 2007 6:14 PM, Marc LEURENT [EMAIL PROTECTED] wrote: Good evening! I was wondering one thing, I'm using freepbx to configure my asterisk server and I have a problem with some inbound calls. When I receive a call to an INVITE sip:[EMAIL PROTECTED] I an set an inbound route! It matches a DID number. How can I route an INVITE sip:[EMAIL PROTECTED] The number only appear in the To: Section. Thanks! Example: With this one, I cannot route it (there is only the number to be reached in the To: section) # U 217.36.112.145:5060 - 192.168.95.235:5060 INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0. Allow: UPDATE,REFER,INFO. Call-ID: [EMAIL PROTECTED] Contact: sip:217.66.118.145:5060. Content-Type: application/sdp. CSeq: 34878212 INVITE. From: 0614740696 sip:[EMAIL PROTECTED];user=phone;tag=02975-US-0223ae6e-67d6c4495. Max-Forwards: 31. To: sip:[EMAIL PROTECTED];user=phone. User-Agent: Cirpack/v4.41c (gw_sip). Via: SIP/2.0/UDP 217.36.112.145:5060;branch=z9hG4bK-744D-33B812. Content-Length: 303. . Whereas with this one I can do it! (there is a number in the INVITE) # U 87.98.202.114:5060 - 192.168.95.235:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0. Via: SIP/2.0/UDP 87.98.202.114:5060;branch=z9hG4bK1fd2c6b4;rport. From: 0158136741 sip:[EMAIL PROTECTED];tag=as25391ca7. To: sip:[EMAIL PROTECTED]. Contact: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Tue, 13 Nov 2007 18:07:00 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 233. . ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Forward on SIP unreachable (network failure)
${DIALSTATUS} will be one of:
- *CHANUNAVAIL* : Channel unavailable (for example in sip.conf, when
using qualify=, the SIP chan is unavailable)
- *BUSY* : Returned busy
- *NOANSWER* : No Answer (i.e SIP 480 or 604 response)
- *ANSWER* : Call was answered
- *CANCEL* : Call attempt cancelled (i.e user hung up before the call
connected)
- *DONTCALL* : Privacy manager don't call
- *TORTURE* : Privacy manager torture menu
- *CONGESTION* : Means Congestion, or anything else (some other error
setting up the call)
Did you test CHANUNAVAIL or CONGESTION ?
Debug DIALSTATUS var for this case using Noop application in dialplan.
On Nov 13, 2007 8:51 PM, Antoine Megalla [EMAIL PROTECTED] wrote:
Hi,
I am trying to implement call forwarding on the event
that my ATA was not
reachable to Asterisk, whether due to registration
timeout, network
interruptions between the ATA and Asterisk, or simply
because the network on
which the ATA resides in unreachable for any reason.
I there a way of implementing such a feature in
Asterisk?
I have implemented CF unconditional, and CF on busy,
CF on unavailable (ring
but no answer) not nothing about CF on (SIP)
unreachable.
Thank you and best regards,
Antoine Megalla.
Never miss a thing. Make Yahoo your home page.
http://www.yahoo.com/r/hs
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Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files
as far as I know, softkey layout is managed by Cisco Call Manager and only available running on skinny protocol. On Nov 13, 2007 2:50 PM, Anciso, Roy [EMAIL PROTECTED] wrote: There is an option to specify a softkey file in SEPmac.cnf.xml. I have an email into our Cisco rep. I'm hoping he can shed some light on this. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Tuesday, November 13, 2007 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. As far as I know (and I might be very wrong), you can't change the soft key configuration of Cisco phones with the SIP Firmware. Maybe you can with Cisco's CallManager - I don't know. Someone PLEASE correct me if I'm wrong because I've been wanting to do this for a year ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Polycom Directory XML via PHP
Hi guys, Any one know about any php or other GUI to manage Polycom Directory XML contacts ? I would like to setup a web page were I manage Directory contacts to provide XML file to feed my polycom phones on Boot with all the contacts. Does any one has already done this? I notice a bash script on wiki, just wondering about php or something else Best regards, Marco Mouta ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to use 1 channel from TE110P for data transmission
Hi guys, I've setup on box with a TE110P and time to time I need to access remote equipment outside of our office and use a data channel. I'm wondering if do I need to buy a POTS line only for this time to time acess or what's the easiest way to do that via my TE110P on asterisk box. I know that is possible data transmission with this Digium Card, I'm wondering how... Any tip any tutorial? Probably someone around the world as already done this before. Best regards, Marco Mouta -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attaching VoiceMails on E-Mails
hi,
The
VoiceMailhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2application
uses
*/usr/sbin/sendmail* to mail voicemail messages to users. This can be any
sendmail-compatible MTA. In practice you can use
Sendmailhttp://sendmail.org/,
Postfix http://postfix.org/, Exim http://www.exim.org/ or any
other MTAhttp://www.voip-info.org/wiki/edit.php?page=Asterisk+voicemail+MTA.
It is recommended to use the default one that comes with your distribution.
I would say if you just create your own sendmail.sh and place it
/usr/sbin/sendmail, asterisk will execute it by default, do not forget to
give permissions for asterisk user to execute it.
On 7/27/07, GNUbie [EMAIL PROTECTED] wrote:
Hello all,
I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to
send the voicemails as attachment to e-mails and delete the voicemails from
my PBX once it has been sent. But, I don't have a running MTA here even on
the PBX itself. I just want to send the e-mails to my GMail account from my
PBX. Can I just use the mail or mailx command to send the e-mail and attach
the voicemail message?
Below are snippets of my voicemail.conf and extensions.conf configuration
files. Please advice whatever you think I need to change with my current
configurations.
Thank you in advance.
GNUbie
- - - s n i p - - -
# cat /etc/asterisk/voicemail.conf
[general]
format=wav49
[EMAIL PROTECTED] ; bogus e-mail address
attach=yes
delete=yes
maxmsg=50
maxmessage=180
minmessage=5
maxgreet=60
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
fromstring=The PBX
usedirectory=yes
emaildateformat=%A, %B %d, %Y at %r
sendvoicemail=yes
emailbody=Hi, ${VM_NAME}!\n\nYou have a new voicemail message from
${VM_CALLERID} attached to this e-mail message.\n\nHave a nice day!\n\nThe
PBX
mailcmd=/usr/bin/exim -t ; not sure about this line
[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
[default]
101 = 11011,GNUbie,[EMAIL PROTECTED]
# grep 10 /etc/asterisk/extensions.conf
exten = 101,1,Dial(Zap/1,20,rt)
exten = 101,2,VoiceMail(101,u)
exten = 100,1,VoiceMailMain(${CALLERID(num)},s)
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Re: [asterisk-users] Best wifi IP phone for asterisk
Siemens GigaSet SL75 On 6/25/07, Michelle Dupuis [EMAIL PROTECTED] wrote: We're looking at a large wifi phone deployment, and we're looking for wifi phones that: 1. Are SIP compliant (Asterisk friendly) 2. Provision capable (ideally TFTP of own MAC address) 3. Industrial quality (no cheap plastic stuff). 4. Well documented (and none of the only telco's get documentation crap) Does anyone have a suggestion? Thanks, MD ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best wifi IP phone for asterisk
i believe www.voipango.de sell them to US On 6/26/07, Nick Seraphin [EMAIL PROTECTED] wrote: On Mon, 25 Jun 2007, Marcus Franke wrote: Benny Amorsen schrieb: MM Siemens GigaSet SL75 The SL75 is DECT, not Wifi. Apart from that, was it really necessary to quote 20 lines and add a ridiculous 15 line disclaimer telling me that I'm not allowed to read the message? There is a GigaSet SL75 WLAN. http://gigaset.siemens.com/shc/0,1935,hq_en_0_122755_rArNrNrNrN,00.html Is this strictly a European phone? I can't find anyone who is selling them in the US... at least not a company I've ever heard of or dealt with before. Tried Amazon.com, voipsupply.com, Tech Data, and 3 pages of Google search results, etc. -- Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX over T1
Hello, You need to setup the following architecture: Asterisk --IAXModemHylafax Then route inboud Fax calls to IAXModem extension outgoing faxes, are sent from user as pdf attachment to Hylafax as email with Destination number in subject, then there is a script hylafax2email pooling sendmail spool every time, that splits all info from email takes the pdf and starts a call via IAXModem transmiting the fax! This links will help U a lot: http://wpkg.org/email2fax/ http://www.hylafax.org/content/Support#Documentation In the past I got all the solution fully working for outgoing and incoming faxes. Best regards, Marco Mouta On 6/22/07, Joe acquisto [EMAIL PROTECTED] wrote: I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines. Have a recently installed Asterisk system, with a dedicated T1 line. (Well, it's really a fonality system). What would I need to do, or where is the reading material, for what I need to do, to convert the Hylafax server to use the T1 line? Reliably. Preferably to use DID's as well. The current FAX works fine, but there is some desire to get rid of the analog lines. Could one add some sort of device in the Asterisk server, to act as FAX extensions, keeping the mainpine on the hylafax? Like a TDM400p with FSX modules? I'm just saying, ya know? I suppose I have to ask fonality, since it's their box? joe a. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ex-Girlfriend Logic in 1.4.4
pleease post your context exactly for the exten 5000 as u have it in live
system.
On 6/19/07, Douglas Garstang [EMAIL PROTECTED] wrote:
I have this in my dialplan…
[general]
static=yes
writeprotect=no
clearglobalvars=no
[start]
exten = 5000,1,Answer
exten = 5000,n,Wait(1)
exten = 5000,n,NoOp(${CALLERID(num)})
exten = 5000,n,Playback(tt-monkeys)
which, when I dial 5000, executes this…
== Parsing '/etc/asterisk/sip_notify.conf': Found
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/5000-0a281f80, ) in new
stack
-- Executing [EMAIL PROTECTED]:2] Wait(SIP/5000-0a281f80, 1) in new
stack
-- Executing [EMAIL PROTECTED]:3] NoOp(SIP/5000-0a281f80, 19256002182)
in new stack
-- Executing [EMAIL PROTECTED]:4] Playback(SIP/5000-0a281f80,
tt-monkeys) in new stack
-- SIP/5000-0a281f80 Playing 'tt-monkeys' (language 'en')
However, when I change the extension match to:
exten = 5000/19256002182,1,Answer
exten = 5000/19256002182,n,Wait(1)
exten = 5000/19256002182,n,NoOp(${CALLERID(num)})
exten = 5000/19256002182,n,Playback(tt-monkeys)
nothing appears on the console and I get no match. You can see the caller
id number is 19256002182 from the NoOp() when it does work. What am I
missing here?
Doug.
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Re: [asterisk-users] which Wifi SIP phones are the good ones
Siemens Gigaset SL75 are just Great! On 6/12/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Tue, 12 Jun 2007, Deepak Naidu wrote: We are planning to buy a wifi SIP phones to work with Asterisk 1.2-18 setup. I would like to get feedback views regarding Linksys WIP300 WIFI IP Phone or any other wifi phones which has been stable. Please check the archives - there was a lot of discussion about just this recently - let's not start it all over again!!! Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any codec passthru mode
so you r sure you have g729 licences installed and ur * is transcoding your RTP streaming? Test the work flow with disallow=all and allow=g729, can be my mistake but I remember to read somewhere on the net any issue about codec negotiating precedence when you use allow=all. good luck On 5/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, My configuration is: USER (connects to) ASTERISK---(connects to)---CARRIER-OUT i want the user preffered codec to pass thru asterisk to carrier-out. what i mean is: USER (user uses g729) ASTERISK---(asterisk should use g729 for dialing out)---CARRIER-OUT instead, this is what happens USER (user uses g729) ASTERISK---(asterisk uses g711u)---CARRIER-OUT How can i force asterisk to use user preffered codec for dialing out so that my asterisk machine saves time by no conversion USER PREFERENCE IS disallow=all allow=g729 CARRIER PREFERENCE IS allow=all Anybody who can help? -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes
FYI, http://www.voip-info.org/wiki/index.php?page=Asterisk+FAQ *Can i install Asterisk on a beowulf cluster?* A cluster can't migrate threads that use shared memory. Asterisk uses that kind of threads.So no, Asterisk wouldn't work on a cluster. *(It might be helpful to know whether anyone has a working load-balanced Asterisk configuration where multiple systems can share the load of an Asterisk environment (IAX2, not SIP) and whether this environment would fail over nicely in the event of downtime!)* On 5/25/07, Sean M. Pappalardo [EMAIL PROTECTED] wrote: Hi there. Just curious if you've checked out Linux clustering software such as OpenSSI ( http://www.openssi.org/ ) and run Asterisk on it? It features a multi-threaded cluster-aware shell (and custom kernel) that will automatically cluster-ize any regular Linux executable (such as the main Asterisk process.) If it works as advertised, it should just be a matter of adding boxes to the cluster to speed up processing. As for Asterisk itself, is it multi-threaded enough to take advantage of 4+ way systems? Sean Pappalardo - This E-Mail message has been scanned for viruses and cleared by SmartMail from Smarter Technology, Inc. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk SIP domain (in LAN or DMZ)?
Hi, In my opinion, you should keep your Asterisk, probably with PSTN Cards, inside your network and just setup an OpenSer or even simpler another Asterisk server on your DMZ. This way you will enable ENUM and URIs for your Clients, and will prevent much better any DoS, intrusion or any other backhole that would let external users places PSTN calls through your server. At the sametime if something goes wrong on outside world, your Lan VoIP going will be kept 99,99% fully functional and let you make and receive calls through PSTN. Good Luck, Marco Mouta Ps. Qualquer coisa apita:) On 5/10/07, Joao Pereira [EMAIL PROTECTED] wrote: Hello I want to use Asterisk to implement a SIP Domain allowing my clients to do URI dialing and receive calls from the Internet through URIs and ENUM. My question is, should I put my Asterisk outside the firewall (in the DMZ) to allow connections to the Internet? Or should I have it inside my local network and put a SIP Proxy (like Openser) in the DMZ to implement the SIP domain? Thanks Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dialing next extension only if first is busy?
Based on my experience I would say that using ${DIALSTATUS} variable would
be the most common way to do it...
On 4/23/07, Daniel Pittman [EMAIL PROTECTED] wrote:
G'day.
I am having reasonable success getting Asterisk 1.4.2 running and doing
what I want, but I can't figure out one particular idiom that I want:
There are a few situations where I want to have Asterisk push a call
through to the first available transport on a list, such as:
I have two SIP ports attached to one local (two port) analog phone
system. I want to ring line 1 for the first call, line 2 for the second
call and go to voicemail for the third and subsequent.
I can't work out the best way to express that.
Using Dial(SIP/line1SIP/line2) will ring both lines at the same time
which is not really what I want.
Using two sequential Dial() commands into the extension will ring the
lines one after the other -- even if it times out on the first line,
which is again not what I want.
At the moment my best guess is that I need to use the DIALSTATUS
variable and implement the fail-over process based on that. That seems
cumbersome, though -- surely this isn't a terribly uncommon requirement?
Regards,
Daniel
--
Digital Infrastructure Solutions -- making IT simple, stable and secure
Phone: 0401 155 707email: [EMAIL PROTECTED]
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Re: [asterisk-users] Asterisk 1.4 with Digium B410P - Timing problem
did you modprobe ztdummy? On 3/30/07, Administrator TOOTAI [EMAIL PROTECTED] wrote: Hi list, we have a dual Xeon server with 2GB RAM running Debian Etch 2.6.18.4-686 kernel. The server has 2 B410P cards plugged in. No other card. We installed Asterisk 1.4 trunk with zaptel trunk, ran make b410p, the install mISDN1.1.0 (for bug 9064) configure and compile Asterisk with chan_misdn, all is fine. In misdn-init.conf we have added option=1,master_clock. Asterisk is up and running, voicemail, echo test, demo, MOH, everything works well. Now we want to add conferences with meetme. So we load zaptel module who created /dev/zap/channel|ctl|pseudo|timer|transcode. Asterisk is running under asterisk:asterisk,we added this user to dialout group to have the good rights on those files. Problem: we can't open conferences, we always have [Mar 30 10:55:37] WARNING[28302]: chan_zap.c:896 zt_open: Unable to open '/dev/zap/pseudo': No such device or address [Mar 30 10:55:37] ERROR[28302]: chan_zap.c:7631 chandup: Unable to dup channel: No such device or address [Mar 30 10:55:37] WARNING[28302]: app_meetme.c:758 build_conf: Unable to open pseudo channel - trying device [Mar 30 10:55:37] WARNING[28302]: app_meetme.c:761 build_conf: Unable to open pseudo device What's wrong? The B410P being a Digium card should gave us a timing source? Do we forgot to compile some module (no one wxxx from zaptel directory is activated)? More generally, how you get the timing working with B410P cards and 1.4? We tried another way, ztdummy: by loading the module, the meetme application start to work but ... no more audio in all applications (voicemail, meetme, MOH, echo-test, demo,...)! In log we found Mar 29 22:40:36 SrvPhone kernel: Zapata Telephony Interface Registered on major 196 Mar 29 22:40:36 SrvPhone kernel: Zaptel Version: SVN-branch-1.4-r2351 Mar 29 22:40:36 SrvPhone kernel: Zaptel Echo Canceller: MG2 Mar 29 22:40:39 SrvPhone kernel: ztdummy: RTC rate is 1024 Mar 29 22:40:39 SrvPhone kernel: rtc: lost some interrupts at 1024Hz. Mar 29 22:40:39 SrvPhone kernel: Registered tone zone 0 (United States / North America) Mar 29 22:40:39 SrvPhone kernel: rtc: lost some interrupts at 1024Hz. Mar 29 22:41:10 SrvPhone last message repeated 1514 times Mar 29 22:42:11 SrvPhone last message repeated 3050 times Mar 29 22:42:57 SrvPhone last message repeated 2348 times Removing ztdummy give audio back but no more meetme :-( Before opening a bug, we would like to know if some of you have a similar working setup with B410P and meetme. Thanks for your feedback. -- Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER vs Asterisk?
Only with Asterisk you can handle it, but of course it depends on your requirements on scalability and redundancy needed. How many agents? How many diferent locations? SIP trunk to your telco or PSTN ? Remote Agents at home? Post more details on your requirements and I believe there are so many experienced users in this list all around the world that you will have good tips here. On 3/23/07, David Anderson [EMAIL PROTECTED] wrote: We're going to be setting up Asterisk at our data center, as well as our call center locations via an optical fiber point to point connection. Is it best to have the servers communicate to eachother via SIP using SER, or just use the Asterisk functions? Thanks, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial(Local/[EMAIL PROTECTED])?
Hi, This is a tool that allows you at any time and any place of your Dialplan or Dialout Call file to dial a specific extension at a specific context, even if you are not currently in the specific context. example: you are at [from-internal] context and you can say: [from-internal] exten= 1234,1,Dial(Local/[EMAIL PROTECTED]) ; Dialing in VIP services, another context, without a GOTO or something else... exten= 1234,2,hangup This is just a simple example, pls take a look on dialout call files and for sure you will notice how useful this local could be. Hope it helps, Best regards, MoutaPT On 3/19/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Rizwan Hisham wrote: I dont understand the syntax of the dial application when used like this: Dial(Local/[EMAIL PROTECTED]) i want to know what is this Local doing instead of Tech like SIP, IAX, H323? SIP/200 would dial a device (the SIP user 200) whereas Local/200 dials the extension 200 in your dialplan (in context longdistance). Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap Load/Stress scripts?
take a look on Originate command for Asterisk manager interface to get web page generating calls between the two boxes. Easier I believe is to use SIPp to be used as an UAC that starts dialing to your box1 and in the dialplan of this box make a dial for a Zap channel on Box2. You need to compile sipp with media streaming and authentication or if you just want first to test you may provide an extension named service in the context defined in general section of your sip conf for external calls coming to your asterisk server without authentication: http://sipp.sourceforge.net/doc/reference.html#Installing+SIPp - *With PCAP playhttp://sipp.sourceforge.net/doc/reference.html#pcapplayand without authenticationhttp://sipp.sourceforge.net/doc/reference.html#authenticationsupport *: # gunzip sipp-xxx.tar.gz # tar -xvf sipp-xxx.tar # cd sipp # make pcapplay - *With PCAP playhttp://sipp.sourceforge.net/doc/reference.html#pcapplayand authenticationhttp://sipp.sourceforge.net/doc/reference.html#authenticationsupport *: # gunzip sipp-xxx.tar.gz # tar -xvf sipp-xxx.tar # cd sipp # make pcapplay_ossl Example: - Sipp being used as a SIP user agent Client: - Call Duration 1ms - Dialing Calls with RTP using ulaw ./sipp -sf uac_pcap.xml -d 1 192.168.34.6 -trace_err Where this IP is my * . On 2/1/07, Mik Cheez [EMAIL PROTECTED] wrote: Use auto dial. You can have as many calls as you wish. http://www.voip-info.org/wiki-Asterisk+auto-dial+out Porier, Jeremy M. wrote: Are there any scripts out there that would help me stress test two boxes that are setup back to back with 4 PRI connections? We're having problems with Sangoma cards w/ PCI-e on HP DL385 G2 hardware and I'm tired of testing them in a production environment. As Sangoma provides firmware updates (and various other shots in the dark) I'd like to be able see if the problem is fixed in an isolated environment. I just need a way to simulate call volume on 4 t1s. Thanks, Jeremy Porier Senior Director of IST Colorado Christian University [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk forgetting about client registration or Polycom phone forgetting to register?
check register expiration on polycom , probably is higher than 3600 sec (default on asterisk) , so after this 3600 , imagine polycom as an expire of 6000sec, there's a gap of 2400sec that polycom isn't registred! On 12/10/06, C F [EMAIL PROTECTED] wrote: While what you say might/should help, it doesn't fix the problem. Additional information, since posting this question, till now, everything worked fine, since it wasn't a working day and only 3 manager sessions are open to asterisk, I'm suspecting that it has to do either if there are lots of phone calls going on, or when the manager has more than 3 active connections. On 12/8/06, Henry J. Cobb [EMAIL PROTECTED] wrote: I'm having trouble with Polycom 501 phones that asterisk forgets how to reach them. ... host=dynamic We've found much better results with the static IP here. Can you try this? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to exit from console?
Try safe_asterisk , for an easy way to start asterisk in background, and then connect with asterisk process running asterisk -rx Now you can use exit, and by the way you may look on wiki diferent ways to run asterisk. On 1/23/07, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI exit No such command 'exit' (type 'help' for help) *CLI quit No such command 'quit' (type 'help' for help) *CLI Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. Any ideas? Thanks, Rudolf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan constructions suggestions?
I don't know about SNOM, but with Xlite Softphone you can have the SoftPhone internal dialplan. Ex. [29];match=1;pre=0; this adds a Zero to every nine digits number s I dial begining with 2 or 9 , this has nothing to do with asterisk, is VoiP phone dialplan. So you can tell to the softphone that when you dial a specific pattern like 9 the Softphone should add an extra 9 in the beginning. This helped me out to import all my contacts from Outlook without having to Add a 0 in the begining of all of them. Hope this help On 1/23/07, Ed W [EMAIL PROTECTED] wrote: Can I ask for some advice on dial-plan construction please I have setup my dialplan to use 9 to get a zap trunk, leaving everything else for internal extensions. However, this creates a problem in that my callerid is correct, but doesn't work to re-dial the incoming caller. So if I simply click missed calls on my Snom phone and hit redial then it tries to dial an internal extension. So I then setup Asterisk to add a 9 to the incoming callerid for all calls which come via the Zap trunk, but now this creates some issues with applications like Snapanumber and perhaps HudLite, which are trying to map the caller ID to numbers in the addressbook (and I don't really want my internal Outlook address books to have everyone listed with a 9 in front of their number) How are others handling this? I have considered simply dropping the prefix digit and working around any clashes in internal and external numbers (not very hard). Grateful for any thoughts Ed W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Audio for Extension to Extension
enable rtp debug in your asterisk CLI and check if there's traffic passing. Would be a first approach I think. On 1/23/07, Tim Panton [EMAIL PROTECTED] wrote: On 22 Jan 2007, at 07:28, Troy - Purple Oranges wrote: I am at a loss, I can terminate and receive calls via any of my providers with both IAX and SIP. I use GSM, G729a, and ulaw for those carriers. If I make an extension to extension call - there is no audio at all in either direction. All my extensions are set to use G729a (I have tried changing that though to see if it would fix it). I am fairly sure it is not a transcoding issue - as the server transcodes for the inbound/outbound calls. You really need to tell us more! At a pure guess however I'd say you have SIP extensions with canreinvite set to true. Your internal network however does not permit rtp traffic between the handsets. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stress-test realtime voicemail with sipp
As far as I know: You need to compile sipp with media streaming and authentication or if you just want first to test you may provide an extension named service in the context defined in general section of your sip conf for external calls coming to your asterisk server without authentication: http://sipp.sourceforge.net/doc/reference.html#Installing+SIPp - *With PCAP playhttp://sipp.sourceforge.net/doc/reference.html#pcapplayand without authenticationhttp://sipp.sourceforge.net/doc/reference.html#authenticationsupport *: # gunzip sipp-xxx.tar.gz # tar -xvf sipp-xxx.tar # cd sipp # make pcapplay - *With PCAP playhttp://sipp.sourceforge.net/doc/reference.html#pcapplayand authenticationhttp://sipp.sourceforge.net/doc/reference.html#authenticationsupport *: # gunzip sipp-xxx.tar.gz # tar -xvf sipp-xxx.tar # cd sipp # make pcapplay_ossl Example: - Sipp being used as a SIP user agent Client: - Call Duration 1ms - Dialing Calls with RTP using ulaw ./sipp -sf uac_pcap.xml -d 1 192.168.34.6 -trace_err Where this IP is my * . Hope this helps, Plse provid some feedback. I would like also to learn from community how to understand Load average results with Top command while incrementing calls dial from sipp to asterisk, and how to determine max calls on Asterisk. This max calls is defined when Sipp calls to * starts being discarded? Best regards, Marco Mouta On 1/23/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: We are in the process of implementing realtime voicemail. I was wanting to stress-test the system to see if or when it would fall over. Is it possible to use sipp to create say 250 calls, each of which leaves a message in the voicemail ? My dialplan is currently [default] exten = stress,1,Answer() exten = stress,2(vm),Voicemail(|su) exten = stress,3,Hangup() however, if I use sipp to test this, I get [Jan 23 14:43:51] WARNING[22782]: app.c:599 __ast_play_and_record: No audio available on SIP/sipp-b7c274b0?? I suspect that's because sipp itself is not sending audio. Is there any tricks I can do in the dialplan to get an extension to answer sipp and then send it to voicemail, but play some audio for the voicemail ? Thanks. Julian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with rxfax
I would install Hylafax(opensource too) with Asterisk via IAXModem, it worth! You can keep all the features, hylafax running in same server or a separated one, and IAXmodem will be a Modem on Hylafax, and a simple extension on you *. It works great for me, and there are more users using this architecture, you can setup as much IAXModem as your servers can handle, so it's very scalable. Best regards, Marco Mouta On 1/22/07, Ardjan Zwartjes [EMAIL PROTECTED] wrote: Dear list, The company I'm working for is trying to use app_rxfax to receive faxes on an Asterisk machine. Our initial tests looked very promising, but unfortunately we've encountered some problems. We've been trying to solve these problems for quite some time now, but we're running out of options. So I really hope that somebody can give some help here. Basically our set-up is this: We have an Asterisk server (version 1.2.7) with an ISDN trunk (Sangoma A104D), we've configured asterisk to run rxfax on a specified extension. Originally we started out with spandsp 0.0.2pre26 and the original app_rxfax for spandsp 0.0.2. Some of the faxes were coming in perfectly, but soon we noticed that quite often there were substantial pieces of the fax missing in the resulting tif file. We've tried the following to solve these problems: - We've checked the timing settings for the ISDN trunk, these seem to be ok. - We've tried several versions of libtiff (currently we are using 3.7.2). - We've tried using 0.0.3 versions of spandsp (since we're using asterisk 1.2.7 we had to modify app_rxfax.c to work). - We've created a custom dialplan application to disable the echo cancellation on the isdn channel on which the fax is received. - We've tried various settings for t30_set_supported_compressions, t30_set_supported_image_sizes, t30_set_supported_modems and t30_set_supported_resolutions (I must confes that I didn't really know which settings to use here, but we have tried a lot of them). - We've tried several fax machines to send the faxes, ranging from simple fax-modems to large dedicated fax machines. But still a lot of faxes give problems, either the tif is missing large portions, or the fax isn't received at all. At the bottom of this mail are 2 examples of the logging when it goes wrong. I really hope that somebody can give a few pointers. Thanks in advance, Kind regards, Ardjan Zwartjes, Telecats === Example 1 = Jan 19 14:49:14 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:14 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:17 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 4 Jan 19 14:49:17 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:19 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 8 Jan 19 14:49:19 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:20 DEBUG[24218] app_rxfax.c: FLOW FAX Switching from V.29 + V.21 to V.29 (-22.18dBm0) Jan 19 14:49:21 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:21 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:23 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 8 Jan 19 14:49:23 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:23 DEBUG[24218] app_rxfax.c: FLOW FAX Switching from V.29 + V.21 to V.29 (-17.84dBm0) Jan 19 14:49:38 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 4 Jan 19 14:49:38 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:39 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:39 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:41 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 8 Jan 19 14:49:41 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:42 DEBUG[24218] app_rxfax.c: FLOW FAX Switching from V.29 + V.21 to V.29 (-19.13dBm0) Jan 19 14:49:48 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 4 Jan 19 14:49:48 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: == Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Pages transferred: 1 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Image size: 1728 x 1192 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Image resolution8037 x 7700 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Transfer Rate: 9600 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Bad rows0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Longest bad row run 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Compression type3 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Image size (bytes) 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: == Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 1 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c
[asterisk-users] Why app_rx and app_tx when we have IAXModem and Hylafax and hy-email2fax? Should we reinvent the wheel?
I'm just wondering here, in our community... Why not HylaFax for Fax solutions? I'm not saying bad about the big efforts on app_tx and app_rx with spandsp. But as me and many other asterisk users, we find out that Hylafax+IAXModem+Asterisk is very reliable. And from what i've been testing and reading, Hylafax is an extremely powerful: HylaFAX is an enterprise-class system for sending and receiving facsimiles as well as for sending alpha-numeric pages. The software is designed around a client-server architecture. Fax modems may reside on a single machine on a network and clients can submit an outbound job from any other machine on the network. Client software is designed to be lightweight and easy to port. HylaFAX is designed to be very robust and reliable. The fax server is designed to guard against unexpected failures in the software, in the configuration, in the hardware and in general use. HylaFAX can support multiple modems and a heavy traffic load. http://www.hylafax.org/content/Main_Page Using this architecture and bash script hy-email2fax, i've been able to setup Email2Fax and Fax2Email. Of course, we the community, could improve and simplify the integration and documentation, as well as the process handled currently by hy-email2fax: http://wpkg.org/email2fax/index.php/Download Is a bad idea to bet on this solution? My apologies, if someone thinks that i'm complaining about app_tx and app_rx, that's not the case! What I mean is that the Real Enterprise Fax Server is not Asterisk, but some one had done it already and it's widely used Hylafax... Please let me know if i'm missing something on this email. Best regards to this great Community, Marco Mouta dCAP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to limit IAX calls
Take a look on: Dialplan applications: GetGroupMatchCount([EMAIL PROTECTED]) SetGroup([EMAIL PROTECTED]) Using this two applications you can deploy a max calls control inside your dialplan! check this too: http://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup Hope it helps On 1/19/07, Barzilai Spinak [EMAIL PROTECTED] wrote: The SIP channels have a call-limit parameter (which is badly documented and I haven't tested yet) How can I have the same behaviour for IAX channels? I can't see anything related to it. Ah, I'm using Asterisk 1.2.13... maybe there is something in the 1.4 versions... but I can't change to 1.4 right now because of MFC/R2 BarZ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-941 (and others ) Transmit Sound Quality
Hi guys, I got also problems with SPA 941 and 942, pour sound (a kind of click noise) that when i set volume sound lower almost can't notice, but still exists. I also notice on SIP to SIP calls , echo that could only be justified by Handsets hardware quality. When i make calls using Xlite with Plantronics DSP 400 USB micro and headset everything works like a charm. I've been told, by some one with longer experience with CISCO phones 7960 that if some one try to just replace in the handset the microphone inside with one form a cheaper traditional phone will get this VoIp hardphone working perfect. But in my case i didn't try that. If someone has a SPA942 on their own lab and can try this without damaging the phone would be nice info to share, I believe! Best regards, Marco Mouta On 1/17/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: I too seem to have the same problem, dont know about poor quality but its certainly not loud enough, I have to put my mouth to the microphone, otherwise the other end reports they cannot hear me. This does however seem to do a good job to cancel out the background noise In the SIPura setup change the packet size from .3 to .2. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] two-level administration tool for Asterisk (reposting)
Freepbx GUI let's you create different administrators with different permissions! On 1/17/07, Kate Kretz [EMAIL PROTECTED] wrote: I like the idea of Virtual PBX, but I don't like python language. Are there other implementations ? I'd like some java or php thing. On 1/16/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Jan 16, 2007 at 10:18:05AM +0500, Kate Kretz wrote: Dear Sirs, let me repost my question again, probably the last one was lost in a huge amount of messages during weekend. I'm actually looking for web-based tool which can do two level of administration: 1) high level, Administrators, can create domains 2) lower level, Users, can manage extensions within certain domain. much like asterisk2billing. so, I want users to manage their things within Asterisk not affecting other users. http://destar.berlios.de/ . What you call domain is called there virtual pbx. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto: [EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] two-level administration tool for Asterisk (reposting)
My mistake Tzafrir, you are right! On 1/17/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Jan 17, 2007 at 11:47:35AM +, Marco Mouta wrote: Freepbx GUI let's you create different administrators with different permissions! But can you separate the permissions by context/domain? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Has been working for 9 Months - Very Very StrangeI cannot dial specific extensions from my dialplan - NOT ACONTEXT PROBLEM!!
with tcpdump i could notice that invites didn't reach my * server. After Rebooting Lan's Firewall CheckPoint problem solved. On 1/12/07, Steven [EMAIL PROTECTED] wrote: Is there a local dialplan on the phone? Maybe these phones were recently upgraded or reset to factory and lost the 4XXX dialplan. That is where I would start. -- -- Steven http://www.glimasoutheast.org Marco Mouta [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX - BROKEN but not for every account. Also I notice that for SIP accounts that can't dial 4XXX they can dial *98 and PSTN calls, and yes they are all in the same context since April 2006! SIP to PSTN - OK SIP to IAX - OK This is a graph from ethereal: Dialing 4214, my own SIP extension! |Time | 192.168.34.26 | XXX.XXX.XX.XX | |11,219 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | |11,721 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | |12,727 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | |14,739 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | |18,762 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | Dialing *98 to check voicemail: 2|21,882 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | 2|21,884 | 407 Proxy Authentication Required |SIP Status | |(2752) -- (61414) | 2|21,886 | ACK | |SIP Request | |(2752) -- (5060) | 2|21,990 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | 2|21,991 | 100 Trying| |SIP Status | |(2752) -- (61414) | 2|21,997 | 200 OK SDP ( g711A GSM g711U telephone-event) |SIP Status | |(2752) -- (61414) | 2|22,034 | RTP (g711U) |RTP Num packets:116 Duration: 2.315s ssrc:490185229 | |(42576) -- (18670) | 2|22,208 | ACK | |SIP Request | |(2752) -- (5060) | 2|23,025 | RTP (g711U) |RTP Num packets:75 Duration:1.484s ssrc:1496378340 | |(42576) -- (18670) | 2|24,523 | BYE | |SIP Request | |(2752) -- (5060) | 2|24,525 | 200 OK| |SIP Status | |(61413) -- (5060) | 2|25,026 | BYE | |SIP Request | |(2752) -- (5060) | 2|25,027 | 200 OK| |SIP Status | |(61413) -- (5060) | Also I notice, with SIP debug peer 4214 on * CLI , that when i dial from my sip phone 4XXX numbers, nothing seems to reach the asterisk Server! I hope someone can point me out where is the problem! This server has only sip extensions. P4 - 1G RAM wiht TE110P with weekly reboot. Best regards, Marco Mouta -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo
Re: [asterisk-users] authentication issue!
You may use astdb for this.
Just set an entry on AstDB with user password and then for every outgoing
call prompt an audio to introduce password and then check if it exists on
AstDB.
User may be the caller ID and the pass is introduced by DTMF.
Then you may use a GOTOIF to allow or not Outgoing Call.
Other choice could be easier:
There are other ways with Authenticate() application
using as an argument for the Authenticate Application a variable defined on
every SIP or IAX account with setvar=mypasswd=1234 ( in sip.conf or iax.conf)
then you may put in your dialplan (extensions.conf) something like:
exten=_X,1,Authenticate(${mypasswd}) ; where mypasswd is a variable
that exist on every sip account definition
exten=_X,2,
http://www.voip-info.org/wiki-Asterisk+Database
http://www.voip-info.org/wiki-Asterisk+cmd+authenticate
Note this are just ideas, you must test it.
You may use Show application Authenticate and Show function DB_EXISTS within
Asterisk CLI to get a deeper understanding of all this functions and
applications
On 1/13/07, William Piper [EMAIL PROTECTED] wrote:
Try www.asterisk2billing.org
On 1/11/07, Pablo Bullian [EMAIL PROTECTED] wrote:
Hi,
I have an issue with the authentication for the outgoing calls.
What I want is to give every user a different password, that they must
enter everytime they make an outgoing call.
What are my possibilities? and can u show me an example please?
Thanks a lot.
--
'May the source be with you'
Pablo E. Bullian
Network Administrator
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[asterisk-users] Has been working for 9 Months - Very Very Strange I cannot dial specific extensions from my dialplan - NOT A CONTEXT PROBLEM!!
Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX - BROKEN but not for every account. Also I notice that for SIP accounts that can't dial 4XXX they can dial *98 and PSTN calls, and yes they are all in the same context since April 2006! SIP to PSTN - OK SIP to IAX - OK This is a graph from ethereal: Dialing 4214, my own SIP extension! |Time | 192.168.34.26 | XXX.XXX.XX.XX | |11,219 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | |11,721 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | |12,727 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | |14,739 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | |18,762 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | Dialing *98 to check voicemail: 2|21,882 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | 2|21,884 | 407 Proxy Authentication Required |SIP Status | |(2752) -- (61414) | 2|21,886 | ACK | |SIP Request | |(2752) -- (5060) | 2|21,990 | INVITE SDP ( BV32 BV32-FEC g711U iLBC g711A GS...elephone-eve) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED]:5060 | |(2752) -- (5060) | 2|21,991 | 100 Trying| |SIP Status | |(2752) -- (61414) | 2|21,997 | 200 OK SDP ( g711A GSM g711U telephone-event) |SIP Status | |(2752) -- (61414) | 2|22,034 | RTP (g711U) |RTP Num packets:116 Duration:2.315s ssrc:490185229 | |(42576) -- (18670) | 2|22,208 | ACK | |SIP Request | |(2752) -- (5060) | 2|23,025 | RTP (g711U) |RTP Num packets:75 Duration:1.484s ssrc:1496378340 | |(42576) -- (18670) | 2|24,523 | BYE | |SIP Request | |(2752) -- (5060) | 2|24,525 | 200 OK| |SIP Status | |(61413) -- (5060) | 2|25,026 | BYE | |SIP Request | |(2752) -- (5060) | 2|25,027 | 200 OK| |SIP Status | |(61413) -- (5060) | Also I notice, with SIP debug peer 4214 on * CLI , that when i dial from my sip phone 4XXX numbers, nothing seems to reach the asterisk Server! I hope someone can point me out where is the problem! This server has only sip extensions. P4 - 1G RAM wiht TE110P with weekly reboot. Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What would make Asterisk Ignore INVITES?
Hi all, As i have already posted, i'm noticing a strange problem on my * server: It seems to me Asterisk is simply ignoring some of invites sent from my xlite 3.0. If i dial 2XXX numbers, all ok. If i dial 4XXX numbers that aren't accounts on asterisk i get answer from asterisk. If i dial 4XXX numbers that exist on my server nothing happens and i get call failed: Request timeout. Calls from PSTN to this SIP extensions 4XXX work FINE. The context is fine, this was working for long time. suddenly seems to get broken. Hope someone can help me on this. Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Block some number outgoing from joust oneextention
post here your extensions.conf On 1/8/07, Mattias Andersson [EMAIL PROTECTED] wrote: Hi! Unfortunately did this stop Asterisk to register ny phones and trunk. Did I put tit in the wrong place? //Mattias Hi! Exactly what I needed. It was the 209 part that I did not figure put. Thanks! //Mattias At 03:53 2007-01-05, you wrote: exten = _9070X./209,1,NoOP,SORRY CHARLIE exten = _9070X./209,2,Congestion This would block any call from 209 to 070X as long as 9 was your outside digit. I use the NoOP to help me out with the CLI and debugging :) Hope this helps Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mattias Andersson Sent: Thursday, January 04, 2007 5:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Block some number outgoing from joust oneextention Thanks! I can´t rely figure out how to block for only one extension. Eg. Extension 209 need to be blocked from making calls starting with 070 (eg. 9070). Some clues did I get bout would it men a new form-internal-blocked dialplan? Regards Mattias On 04/01/07, C F mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: The easiest way is thru using contexts. On 1/3/07, Mattias Andersson mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all! I am shore someone have writing about it bout I cant find it. I have a extension that I need to block from making expansive mobil calls. Everyone else should be aloud to do the calls. I am shore it is possible to be done sens I had a commercial asterisk based PBX that I did that on. However I have switch to Trixbox because I need some custom functions not supported by the commercial product. I would appreciate all help. Regards Mattias Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by http://Easynews.comEasynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by http://Easynews.comEasynews.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.16.4/615 - Release Date: 1/3/2007 1:34 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Block some number outgoing from joust oneextention
That's what i told you Mattias. On 1/5/07, Mattias Andersson [EMAIL PROTECTED] wrote: Hi! Exactly what I needed. It was the 209 part that I did not figure put. Thanks! //Mattias At 03:53 2007-01-05, you wrote: exten = _9070X./209,1,NoOP,SORRY CHARLIE exten = _9070X./209,2,Congestion This would block any call from 209 to 070X as long as 9 was your outside digit. I use the NoOP to help me out with the CLI and debugging :) Hope this helps Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mattias Andersson Sent: Thursday, January 04, 2007 5:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Block some number outgoing from joust oneextention Thanks! I can´t rely figure out how to block for only one extension. Eg. Extension 209 need to be blocked from making calls starting with 070 (eg. 9070). Some clues did I get bout would it men a new form-internal-blocked dialplan? Regards Mattias On 04/01/07, C F mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: The easiest way is thru using contexts. On 1/3/07, Mattias Andersson mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all! I am shore someone have writing about it bout I cant find it. I have a extension that I need to block from making expansive mobil calls. Everyone else should be aloud to do the calls. I am shore it is possible to be done sens I had a commercial asterisk based PBX that I did that on. However I have switch to Trixbox because I need some custom functions not supported by the commercial product. I would appreciate all help. Regards Mattias Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by http://Easynews.com Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by http://Easynews.comEasynews.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.16.4/615 - Release Date: 1/3/2007 1:34 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] yet another faxing issue (outbound only, via ATA)
Hi all, I was having a similar issue, using TE110P from Digium all incoming faxes were detected and correctly received. When trying to send outbound faxes, they all get broken... I do believe it may be related with Echo Cancel enabled on my Zapata.conf, any ways i've set also fax detect for inbound and outbound on zapata, but that stills diferent from the receiving model as it relies on NVfaxdetect to detect. After many trials, i setup an architecture with another Server Running Hylafax and IAXmodem registring on my * Box and i just get out of troubles. It's perfect sending and receiving faxes with notifications and everything else, Hylafax + IAXmodem and Asterisk are working like a charm. I must say that we don't send too many faxes per day, but until now no problems! And yes didn't change anything on Zapata config or something else on Asterisk Box, i just added IAX account registred my Hylafax IAXmodem there and Voilá :) On 1/3/07, Thomas Kenyon [EMAIL PROTECTED] wrote: Bill Gibbs wrote: My next step is to connect the fax machine to a Wildcard X100P. Check to see if there is Echo cancellation in the SPA-1001, and if so turn it off. If there is an adaptive Jitter buffer on the SPA-1001, try changing it to a fixed one (probably no more than 40ms). Why would you connect a fax machine to an X100P, aren't they FXO cards? Have you tried terminating to a VOIP provider? (to see if the problem is with the ATA). Here I use a fax machine connected to a CS6220 which is connected to the asterisk box and terminates with a TDM400P card (so a completely different arrangement). Any other suggestions? Black magic? Voodoo? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM loses server registration
Hi Joao, I'm not very experienced with SNOM, but have you though about providing fix IP for you VoIP hardphones? That way you could avoid the registration problem. At least while you don't get your final solution. Hope it helps, MoutaPT On 1/3/07, Joao Pereira [EMAIL PROTECTED] wrote: Hello to all When my SNOM (300 or 320) loses Internet connectivity, it loses its Asterisk registration (ok, thats normal). But when the phone is back online, he doesn't try to register in Asterisk. I believe this happens to avoid flooding the private LANs when the Internet link is lost but the problem is that the phones don't try to re-register in the future Sometimes it stays 2 hours without registering to Asterisk. When this happens, the only solution is to reboot it (and hear the users complains) :( How can I avoid this? How can I reduce the time to re-register in SNOM 300 or 320 ? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Block some number outgoing from joust one extention
Hi Mattias, add this to your dialplan: exten= _/CALLERIDNUMBER,1,Hangup() ; Basically you are doing a pattern match with callerid match on your first priority! ; You may keep your remaining dialplan, no changes needed Pls Give me some feedback Best Regards, MoutaPT On 1/3/07, Mattias Andersson [EMAIL PROTECTED] wrote: Hi all! I am shore someone have writing about it bout I cant find it. I have a extension that I need to block from making expansive mobil calls. Everyone else should be aloud to do the calls. I am shore it is possible to be done sens I had a commercial asterisk based PBX that I did that on. However I have switch to Trixbox because I need some custom functions not supported by the commercial product. I would appreciate all help. Regards Mattias Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Presence issues with Got SUBSCRIBE for extensions without hint. Please add hint to s
Are you sure there are no VoIP Phone users with Eyebeam or even polycom requesting SUBSCRIBE for other extensions? It happened to me, that users on my network were adding Subscribe for PSTN numbers that aren't even extensions on my * server. On 12/29/06, Lorentz Hinrichsen [EMAIL PROTECTED] wrote: Hello all, I have a number of Polycom phones 601's and 430's and I'm seeing: Got SUBSCRIBE for extensions without hint. Please add hint to s to context local-hints in the CLI over and over. I have: [local-hints] exten = 110,hint,SIP/110 exten = 111,hint,SIP/111 exten = 112,hint,SIP/112 exten = 113,hint,SIP/113 exten = 114,hint,SIP/114 The hints seem to be working, however why is it looking for a hint for s - should I define one? Polycom's are running 1.6.7, Asterisk is 1.2.9.1 Thanks in advance wulf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] System Application with java
Does the user who is running asterisk has permissions to execute it? check
you script file permissions.
On 12/22/06, Andre Gustavo Lomonaco [EMAIL PROTECTED] wrote:
Hi,
I created a script named example2.sh which goal is read some text from my
HP Service Desk using an application in java and send this text to the
text2wave application for TTS.
example2.sh
java -Xbatch Example10 | text2wave -f 8000 -o /var/lib/asterisk/sounds/my-
sd.wav
When I execute the script in prompt, everything is ok, but when I use the
system() command in my
extensions.conf it isn´t work, just a small file my-sd.wav is created.
Here my extensions.conf configuration
;testing text2wav
exten = 666,1,Answer
exten = 666,2,system(/root/example2.sh /root/log.txt )
exten = 666,3,system(echo ${SYSTEMSTATUS} /root/log.txt)
exten = 666,4,wait(10)
exten = 666,5,Playback(my-sd)
exten = 666,6,Hangup
And here the logging by Asterisk..
Connected to Asterisk 1.2.13 currently running on fedora (pid = 1951)
Verbosity is at least 3
-- Remote UNIX connection
-- Executing Answer(SIP/lomonaco-0945fd18, ) in new stack
-- Executing System(SIP/lomonaco-0945fd18, /root/example2.sh
/root/log.txt ) in new stack
-- Executing System(SIP/lomonaco-0945fd18, echo SUCCESS
/root/log.txt) in new stack
-- Executing Wait(SIP/lomonaco-0945fd18, 10) in new stac
Any help or tip
Thanks in Advanced
Andre Lomonaco
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Re: [asterisk-users] Trying to forward calls by using the Callee's context as the forward dial context
Hi John,
I would try to use on sip.conf and iax.conf and zapata.conf:
on every user (friend or whatever) defined add this:
[useraccount]
setvar=mycontext=yourcontext
--
This variable will become available for every user, so you just need to use
it in your dialplan in extensions.conf
Noop(User context:$mycontext)
This is just an idea, please give some feedback if it helped and some how
you will test if the forward number is valid or not :)
Probably isn't hard , but is not clear yet for me and i'm busy :)
Best Regards,
Marco Mouta
On 12/15/06, John French [EMAIL PROTECTED] wrote:
I'm simply trying to forward calls to users who have the call forwarding
feature enabled (FWD Status and FWD Ph Number kept in the astDB). The
problem is that I want users to be able to forward calls to numbers that
they would normally be allowed to dial within their own context. (I
don't want a local call only person forwarding to a long dist number,
for example.) I'm able to get the channel context for SIP devices but
not for IAX or Zap Devices. I need some pointers on getting IAXPEER to
work and how to handle getting the ZAP context info. If there's an
easier way, I'm all ears. Thanks.
; #Set Some Variables
exten = s,1,Set(DEVICE=${ARG1}) ;i.e. SIP/johns_phone
exten = s,n,Set(Protocol=${CUT(DEVICE,/,1)}) ;Parse out SIP
exten = s,n,Set(Phone=${CUT(DEVICE,/,2)}) ;Parse out johns_phone
;Stuff omitted for some amout of brevity
; #Make Forward Calls##
; We only want people to be able to forward to numbers they could
normally call
; We'll have to somehow get their dialing contexts from the channel conf
files.
exten = s,n(Forward),NoOp()
exten = s,n,GotoIf($[${Protocol} = SIP]?SIPDev)
exten = s,n,GotoIf($[${Protocol} = IAX2]?IAXDev)
exten = s,n,Goto(ZapDev)
;ok, they are an IAX device so use IAXPEER
exten =
s,n(IAXDev),Set(CalledUsersContext=${IAXPEER(${Phone}:context)})
exten = s,n,Goto(dial_time)
;ok, they are an SIP device so use SIPPEER
exten =
s,n(SIPDev),Set(CalledUsersContext=${SIPPEER(${Phone}:context)})
exten = s,n,Goto(dial_time)
;ok, they are an Zap device so use... Uh.
exten = s,n(ZapDev),NoOp( I have no clue how to get the zap channel's
context...)
exten = s,n(dial_time),NoOp(== Chan Type
${Protocol})
exten = s,n,NoOp(== Chan Name ${Phone})
exten = s,n,NoOp(== Channel User's context
${CalledUsersContext})
exten = s,n,Dial(Local/[EMAIL PROTECTED]/n)
Results at console on verbosity 9:
SIPPEER() Works as advertised when I dial a SIP phone which has been
call forwarded
-- Executing NoOp(Zap/1-1, == Chan Type
SIP) in new stack
-- Executing NoOp(Zap/1-1, == Chan Name
jf_linksys) in new stack
-- Executing NoOp(Zap/1-1, == Channel Users
context longdistance_users) in new stack
-- Executing Dial(Zap/1-1, Local/[EMAIL PROTECTED]/n)
in new stack
IAXPEER() Seems to be broken or I don't know how to use it properly.
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Chan Type IAX2) in new stack
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Chan Name johns_pc) in new stack
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Channel Users context ) in new stack
-- Executing Dial(SIP/jf_linksys-08f20548, Local/5551212@/n) in
new stack
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Re: [asterisk-users] Trying to forward calls by using the Callee's context as the forward dial context
forking CDR could help Ricardo.
On 12/15/06, Ricardo Martins [EMAIL PROTECTED] wrote:
Hi John, I´m very interested into this call forwarding capabilities and
I solved this problem filtering on the web-script (in my case, php) the
number the user can intert on the database. (I know it´s not an asterisk
solution).
There is an issue that I couldn´t handle. When I forward the call, I
want to charge the user that the call was made FOR. How are you dealing
with that? Going direct to the point, I just need to know - a tip would
be apreciated either - how to translate/replace the FROM field of the
forwarded call.
Rgds, Ricardo.
John French wrote:
I'm simply trying to forward calls to users who have the call forwarding
feature enabled (FWD Status and FWD Ph Number kept in the astDB). The
problem is that I want users to be able to forward calls to numbers that
they would normally be allowed to dial within their own context. (I
don't want a local call only person forwarding to a long dist number,
for example.) I'm able to get the channel context for SIP devices but
not for IAX or Zap Devices. I need some pointers on getting IAXPEER to
work and how to handle getting the ZAP context info. If there's an
easier way, I'm all ears. Thanks.
; #Set Some Variables
exten = s,1,Set(DEVICE=${ARG1}) ;i.e. SIP/johns_phone
exten = s,n,Set(Protocol=${CUT(DEVICE,/,1)}) ;Parse out SIP
exten = s,n,Set(Phone=${CUT(DEVICE,/,2)}) ;Parse out johns_phone
;Stuff omitted for some amout of brevity
; #Make Forward Calls##
; We only want people to be able to forward to numbers they could
normally call
; We'll have to somehow get their dialing contexts from the channel conf
files.
exten = s,n(Forward),NoOp()
exten = s,n,GotoIf($[${Protocol} = SIP]?SIPDev)
exten = s,n,GotoIf($[${Protocol} = IAX2]?IAXDev)
exten = s,n,Goto(ZapDev)
;ok, they are an IAX device so use IAXPEER
exten =
s,n(IAXDev),Set(CalledUsersContext=${IAXPEER(${Phone}:context)})
exten = s,n,Goto(dial_time)
;ok, they are an SIP device so use SIPPEER
exten =
s,n(SIPDev),Set(CalledUsersContext=${SIPPEER(${Phone}:context)})
exten = s,n,Goto(dial_time)
;ok, they are an Zap device so use... Uh.
exten = s,n(ZapDev),NoOp( I have no clue how to get the zap channel's
context...)
exten = s,n(dial_time),NoOp(== Chan Type
${Protocol})
exten = s,n,NoOp(== Chan Name ${Phone})
exten = s,n,NoOp(== Channel User's context
${CalledUsersContext})
exten = s,n,Dial(Local/[EMAIL PROTECTED]/n)
Results at console on verbosity 9:
SIPPEER() Works as advertised when I dial a SIP phone which has been
call forwarded
-- Executing NoOp(Zap/1-1, == Chan Type
SIP) in new stack
-- Executing NoOp(Zap/1-1, == Chan Name
jf_linksys) in new stack
-- Executing NoOp(Zap/1-1, == Channel Users
context longdistance_users) in new stack
-- Executing Dial(Zap/1-1, Local/[EMAIL PROTECTED]/n)
in new stack
IAXPEER() Seems to be broken or I don't know how to use it properly.
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Chan Type IAX2) in new stack
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Chan Name johns_pc) in new stack
-- Executing NoOp(SIP/jf_linksys-08f20548,
== Channel Users context ) in new stack
-- Executing Dial(SIP/jf_linksys-08f20548, Local/5551212@/n) in
new stack
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Re: [asterisk-users] send fax by Iaxmodem ?
Hi all, First let me say thank you for Lee Howard, you definitely found my problem on sending faxes! I'm using hy-email2fax to send faxes, and i notice that is there the problem is starting, as the subject of my .eml file contains only the phone number but then some how hy-email2fax is not detecting the ^M at the end of subject. I'm not an used with sed so it will take me some time to understand how it works and correct this. By the way i notice a small isse bug on hy-email2fax available for download on http://wpkg.org/email2fax/files/hy-email2fax -- # # Make a temporary file out of an email # # cat | sed 's/ $//'$EMAILFILE -- I only got Hy-email2fax working fine after changing this to: cat | sed 's/$//'$EMAILFILE Any ways this is working fine only for tests, because as i mentioned above I'm in troubles with a carriage return on email subject, to test Sucessfully this hy-email2fax i hardcoded the variable NEWNUMBER= to my dialed number. After this Hands on I can sucessfully send faxes with Hy-email2fax -- Hylafax---asterisk Sucessfully. But as i mentioned before i need to get ride of ^M on the subject line. Any one can help me on this? Best regards, Marco Mouta On 12/13/06, Lee Howard [EMAIL PROTECTED] wrote: Marco Mouta wrote: Dec 13 11:28:07.51: [ 9242]: DIAL 2079^M Dec 13 11:28:07.51: [ 9242]: -- [9:ATDT2079\r] Dec 13 11:28:16.70: [ 9242]: -- [4:BUSY] Dec 13 11:28:46.70: [ 9242]: MODEM TIMEOUT: reading line from modem Dec 13 11:28:46.71: [ 9242]: MODEM Timeout Dec 13 11:28:46.71: [ 9242]: SEND FAILED: JOB 1 DEST 2079^M ERR Unknown problem In your case BUSY means exactly that, and you should take a look at the Asterisk CLI to get more information as to what busy really means. However, your dialstring terminated by a carriage return (CR or ^M) is problematic, too, because it essentially instructs HylaFAX to ignore all responses after ATDT2079 except for OK and then proceed from there. Basically you just need to get rid of that terminating carriage return. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to temporarily unload modules.
/etc/asterisk/modules.conf On 12/13/06, Angel Heart [EMAIL PROTECTED] wrote: Hi, In what Asterisk file can I edit for me to temporarily unload such modules. But later I woudl still be able to load them. Thanks Angel -- Cheap Talk? Check outhttp://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.comYahoo! Messenger's low PC-to-Phone call rates. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] send fax by Iaxmodem ?
Hi Guys, I'm using Asterisk with Hylafax to send and receive faxes, currently only receinving with success. When sending i get this: Dec 13 11:28:07.51: [ 9242]: SESSION BEGIN 00157 03510212079 Dec 13 11:28:07.51: [ 9242]: HylaFAX (tm) Version 4.3.1 Dec 13 11:28:07.51: [ 9242]: SEND FAX: JOB 1 DEST 2079^M COMMID 00157 DEVICE '/dev/ttyIAX' FROM 'Marco Mouta [EMAIL PROTECTED]' USER root Dec 13 11:28:07.51: [ 9242]: STATE CHANGE: RUNNING - SENDING Dec 13 11:28:07.51: [ 9242]: -- [12:AT+FCLASS=1\r] Dec 13 11:28:07.51: [ 9242]: -- [2:OK] Dec 13 11:28:07.51: [ 9242]: MODEM set XON/XOFF/FLUSH: input ignored, output disabled Dec 13 11:28:07.51: [ 9242]: DIAL 2079^M Dec 13 11:28:07.51: [ 9242]: -- [9:ATDT2079\r] Dec 13 11:28:16.70: [ 9242]: -- [4:BUSY] Dec 13 11:28:46.70: [ 9242]: MODEM TIMEOUT: reading line from modem Dec 13 11:28:46.71: [ 9242]: MODEM Timeout Dec 13 11:28:46.71: [ 9242]: SEND FAILED: JOB 1 DEST 2079^M ERR Unknown problem Dec 13 11:28:46.71: [ 9242]: -- [5:ATH0\r] Dec 13 11:28:46.71: [ 9242]: -- [2:OK] Dec 13 11:28:46.71: [ 9242]: MODEM set DTR OFF Dec 13 11:28:46.71: [ 9242]: MODEM set baud rate: 0 baud (flow control unchanged) Dec 13 11:28:46.71: [ 9242]: STATE CHANGE: SENDING - MODEMWAIT (timeout 5) Dec 13 11:28:46.71: [ 9242]: SESSION END I Must say on the reception side is a normal fax connected to pstn line, and to send fax via Asterisk+Hylafax i've tested TE110P and X100P board. I got few sucess with x100p and couldn't send even one with TE110p Any tips? On 12/13/06, Noc Phibee [EMAIL PROTECTED] wrote: Hi i use now iaxmodem for receive fax and that's work very good with Hylafax ;=) Do you know if we can sent fax using iaxmodem and Hylafax ? when i test: déc 13 13:47:21.12: [13725]: SESSION BEGIN 00014 330426690268 déc 13 13:47:21.12: [13725]: HylaFAX (tm) Version 4.3.0 déc 13 13:47:21.12: [13725]: SEND FAX: JOB 2 DEST 0426690268 COMMID 00014 DEVICE '/dev/iaxmodem1' FROM 'localtest' USER test déc 13 13:47:21.12: [13725]: STATE CHANGE: RUNNING - SENDING déc 13 13:47:21.12: [13725]: -- [12:AT+FCLASS=r] déc 13 13:47:21.12: [13725]: -- [2:OK] déc 13 13:47:21.12: [13725]: MODEM set XON/XOFF/FLUSH: input ignored, output disabled déc 13 13:47:21.12: [13725]: DIAL 0426690268 déc 13 13:47:21.12: [13725]: -- [15:ATDT0426690268\r] déc 13 13:47:21.12: [13725]: -- [11:NO DIALTONE] déc 13 13:47:21.12: [13725]: SEND FAILED: JOB 2 DEST 0426690268 ERR No local dialtone déc 13 13:47:21.12: [13725]: -- [5:ATH0\r] déc 13 13:47:21.12: [13725]: -- [2:OK] déc 13 13:47:21.12: [13725]: MODEM set DTR OFF déc 13 13:47:21.12: [13725]: MODEM set baud rate: 0 baud (flow control unchanged) déc 13 13:47:21.12: [13725]: STATE CHANGE: SENDING - MODEMWAIT (timeout 5) déc 13 13:47:21.12: [13725]: SESSION END Thanks bye ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskNow console access
me too, i'm trying to add sip users , i click save, it reports successfully saved... but there are no sip accounts created... On 11/29/06, Dumpolid Exeplish [EMAIL PROTECTED] wrote: i had the same problem. the GUI stopped responding to configuration changes. On 11/28/06, James Willing [EMAIL PROTECTED] wrote: Geoff Karl [EMAIL PROTECTED] wrote: I just downloaded and installed the AsteriskNow appliance (http://www.asterisknow.org) . This looks like it has lots of promise. Anyone know what the secret is to being able to actually login to the root console? Yes, as I found out (rather painfully) after the second (or was it third) install, for console access you have to login as 'admin', using the password you entered during the installation. And I agree that it looks promising, though as far as I can tell so far none of the GUI functionality actually works yet. So far, I have been unable to actually get it to commit any changes entered via the GUI and after a few attempts (or an hour or so of running) the GUI generally appears to stop functioning. No response to 'system info' selection, etc... ...but it is 'Beta 1' afterall... B^} -- -jim (Willing) Midwest Connections, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Low beep on voicemail
take a look on Audacity program is opensource and has the option Generate Beep, then just add some Gain as you want... On 12/2/06, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: We've had a few people complain that the beep before leaving a voicemail is not loud enough and too short. Does anybody have a recorded beep that they can share, that is a little louder and a little longer? We've had this box in production for 2+ years, so I hate to mess with the gain on the PRI or anything like that because everything else works fine. I know nothing about recording sounds, and I am sure I could spend a few hours and come up with a suitable version, but I thought I'd ask around before I waste my time trying to figure it out. Thanks in advance. Peder ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Feature Codes won't work
post you features.conf as well as extensions.conf On 11/27/06, Mattias Andersson [EMAIL PROTECTED] wrote: Hi all! I get problem with *11, *12 for instance. The won't work. I get a message that the phone extension can't be fund for *11 and for *12 will I get A Error. Any idea? //Mattias -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls don't arrive for correct number
your problem is that you need to handle this in your dialpan to achieve which DID has been dialed! look for SIPGETHEADER application on asterisk, you shoul look for variable to where it comes the DID On 11/27/06, Frederico Madeira [EMAIL PROTECTED] wrote: I have an asterisk box registering 100 numbers on a voip provider. Numers are: 2546.1000 to 2546.1099 My problem is that every incoming call arrived to number 2546.1099 that is the last number to register on voip provider. The correct is call arrive in destination number. See this exaple: I call to 2546.1000. -- Executing Dial(SIP/25461099-08738060, Zap/g1/3000) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/3000 -- Zap/1-1 is proceeding passing it to SIP/25461099-08738060 -- Zap/1-1 is ringing -- Zap/1-1 answered SIP/25461099-08738060 -- Hungup 'Zap/1-1' == Spawn extension (default, 25461000, 1) exited non-zero on 'SIP/25461099-08738060' How i solve this problem ?? See parts of my sip.conf register=25461000:[EMAIL PROTECTED]/25461000 register= 25461001:[EMAIL PROTECTED]/25461001 register=25461002:[EMAIL PROTECTED]/25461002 register= 25461003:[EMAIL PROTECTED]/25461003 . . . register=25461099:[EMAIL PROTECTED]/25461099 [provider-25461000] type=friend context=default secret= username=25461000 host=sip.provider.com fromuser=25461000 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes [provider-25461001] type=friend context=default secret= username=25461001 host=sip.provider.com fromuser=25461001 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes [provider-25461002] type=friend context=default secret= username=25461002 host=sip.provider.com fromuser=25461002 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes . . . [provider-25461099] type=friend context=default secret= username=25461099 host=sip.provider.com fromuser=25461099 fromdomain=sip.provider.com nat=yes insecure=very canreinvite=no qualify=yes -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfers in SER + Asterisk architecture
Hi Ricardo, Could you post a specific example where your problem happens. That way would be easier for me to try to help you on this. Does asterisk is registred into SER , or you have trust based relationship between them? On 11/23/06, Ricardo Carvalho [EMAIL PROTECTED] wrote: Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem is that when the REFER message (that carries the Refer-To number to whom the call should be transferred) gets to Asterisk, it replies with a 404 Not Found message, and the Call Transfer isn't established! Any ideas on how can I solve this problem? Thanks in advance, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfers in SER + Asterisk architecture
do you have created Asterisk views to SER database? Are you using sip realtime on asterisk? please post your extensions.conf. By the way, I'm Portuguese:) Qualquer coisa manda mail pode ser q consiga ajudar. On 11/24/06, Ricardo Carvalho [EMAIL PROTECTED] wrote: Hi Marco, Ser has IP of Asterisk server in his trusted table, Asterisk isn't registered in Ser. When Ser needs to use Asterisk, it simply rewrites the IP destination with Asterisk's IP, and routes them to him. For example, here's one failed attempt in transferring a call PSTN - VoIP - VoIP: PSTN Asterisk Ser phone_A phone_B |INVITE| | | | | -- | | | | | 100 Trying | | | | | --- | | | | | | INVITE| | | | | -- |INVITE | | | | | --- | | | | |100 trying | | | 100 trying | --- | | | 100 trying | --- | 180 Ringing | | | -- | 180 Ringing | --- | | | 180 Ringing | -- | | | | -- | | | | | ACK | | | | | --- | ACK | | | | | --- | ACK | | | | | --- | | | | RTP | | | | == | | | | | | | | | | REFER | | | | REFER| --- | | | | -- | | | | | 404 Not Found | | | | | --- | 404 Not Found | | | | | -- | | | | | | | In this example, phone_A answers the PSTN originated call, and wants to transfer the call to phone_B. A REFER message is than routed backwards to Asterisk, and he replies with those 404 Not Found messages. Phone_B never gets called! Should Asterisk be registered in Ser so this can work properly? How can that be done? Thanks, Ricardo. Marco Mouta wrote: Hi Ricardo, Could you post a specific example where your problem happens. That way would be easier for me to try to help you on this. Does asterisk is registred into SER , or you have trust based relationship between them? On 11/23/06, *Ricardo Carvalho* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem is that when the REFER message (that carries the Refer-To number to whom the call should be transferred) gets to Asterisk, it replies with a 404 Not Found message, and the Call Transfer isn't established! Any ideas on how can I solve this problem? Thanks in advance, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE110P and TDM400P
try this, pls give some feedback ### /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 fxsks=1-4 bchan=5-19,21-35 dchan=20 loadzone = us defaultzone=us ### On 11/22/06, Lincoln Zuljewic Silva [EMAIL PROTECTED] wrote: This is the scenarios: 1 - ### /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=5-19,21-35 dchan=20 fxsks=1-4 loadzone = us defaultzone=us ### modprobe wcte11xp ZT_CHANCONFIG failed on channel 32: No such device or address (6) FATAL: Error running install command for wcte11xp 2 - ### /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=5-19,21-35 dchan=20 fxsks=1-4 loadzone = us defaultzone=us ### modprobe wctdm ZT_CHANCONFIG failed on channel 5: No such device or address (6) FATAL: Error running install command for wctdm 3 - ### /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 fxsks=32-35 loadzone = us defaultzone=us ### modprobe wcte11xpok modprobe wctdmok modprobe wcfxook modprobe wct4xxpok modprobe zaptelok ### /etc/asterisk/zapata.conf [channels] context=corsidian overlapdial=yes immediate=no callprogress=yes busydetect=no switchtype=euroisdn signalling=pri_net channel = 1-15,17-31 group=2 group=1 callgroup=1 pickupgroup=1 signalling=fxs_ks channel = 32-35 ### tail -f /var/log/asterisk/messages Nov 22 15:11:43 ERROR[5524] chan_zap.c: Channel 16 is reserved for D-channel. Nov 22 15:11:43 ERROR[5524] chan_zap.c: Unable to register channel '1-15' Nov 22 15:11:43 WARNING[5524] loader.c: chan_zap.so: load_module failed, returning -1 Nov 22 15:11:43 WARNING[5524] loader.c: Loading module chan_zap.so failed! - Original Message - *From:* Henk Dick [EMAIL PROTECTED] *To:* 'Lincoln Zuljewic Silva' [EMAIL PROTECTED] ; 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com *Sent:* Wednesday, November 22, 2006 4:08 PM *Subject:* RE: [asterisk-users] TE110P and TDM400P I think that you are loading the drivers in the wrong order. You can change the order of loading are first define the E1 followed by the TDM400 Hope this helps, Henk -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Lincoln Zuljewic Silva *Sent:* woensdag 22 november 2006 20:51 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] TE110P and TDM400P Hello all. I have here a TE110P (configured as E1) and a TDM400P (with four X100P - FXS). Both boards are recognized by the operating system as showed above: :08:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b1d9:0003 Flags: bus master, medium devsel, latency 64, IRQ 169 I/O ports at e800 [size=256] Memory at febff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 :08:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 79fe:0001 Flags: bus master, medium devsel, latency 64, IRQ 193 I/O ports at e400 [size=256] Memory at febfe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 The problem is that I cant make the both cards to work together in the same server. Here is my /etc/zaptel.conf: ### fxsks=1-4 loadzone = us defaultzone=us span=1,1,0,ccs,hdb3,crc4 bchan=5-19,21-35 dchan=20 ### When I load the wctdm module, I get this error: ZT_CHANCONFIG failed on channel 5: No such device or address (6). Its sounds like the FXS module its tring to configure the channels 5 to 35 (E1 - ISDN Channels - TE110P card). Anybody already saw this ? Its possible to use this two cards in the same computer ? There is any separator that I can use in zaptel.conf to make the load of the modules dont mistakes itself ? Here is my versions: Debian kernel - 2.6.8 asterisk-1.2.12.1 libpri-1.2.4 zaptel-1.2.11 Thanks Lincoln ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FOP is not displaying all my SIP extensions neither all E1 channels , why?
Hi, I must say that i'm not very used with customization of FOP. I've a box runing Flash Op.Panel, and i notice that the screen is full of buttons from my sip users, as well as Zapata channels. The problem is that i have more Zapata channels as well as SIP users, is there any way to get a scroll on this to display everything? do i need to resize the buttons? For sure someone now how to solve this basic question:) -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Rewriting caller ID from database?
Hi, You can do it using AstDB, just load the database with callerid names and numbers and then include a lookup on database in all incoming calls, so you can override whatever you wanted:) On 11/22/06, Steven [EMAIL PROTECTED] wrote: There are two I can think of. Hoodahek and asterdex (or asteridex) We used hoodahek at first, but now use asterdex(sp?) It has a web interface to enter the new names into. We use it to fixup, corp. cell phones and used to use it for our leagcy PBX extensions. -- -- Steven http://www.glimasoutheast.org Vincent Delporte [EMAIL PROTECTED] wrote in message news: [EMAIL PROTECTED] Hi Most of our customers have generic names like Hospital, so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as Reading Hospital so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager: equivalent of 'show channels'?
take a look at Flash Operator Panel, as far as i know they use AMI , and they also provide real time channel status. On 11/18/06, Michael Collins [EMAIL PROTECTED] wrote: I'm interested in knowing if anyone else has worked around this issue: I have an application that needs to check the status of the calls going through Asterisk about every 5 seconds or so. I don't want to do asterisk –rx 'show channels verbose' at the Linux command line 12 times per minute so I am looking at the AMI. I see that there isn't a manager command for 'show channels.' Has anyone come up with an equivalent of 'show channels' using the extant manager commands? If so, could you post how you did it? Thanks! -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A question on ISDN cards... (in the UK)
Beronet cards have 2 or 4 ports are very good.Those guys produced the misdn driver, that is now Digium uses for their new BRI card.www.beronet.comtheir tech support has been very very good. On 11/15/06, Conrad Wood [EMAIL PROTECTED] wrote: On Wed, 2006-11-15 at 11:23 +, Senad Jordanovic wrote: Would anyone like to recommend a good and reasonable quality ISDN card for use in the UK, as after a lot of good results with TDM400P cards with several systems installed now, I need to look at a few ISDN BRI (old business highway about to move to ISDN2) and possibly a single-line PRI (ISDN-30) system. I'd say the most important distinction is the choice between HFC basedISDN cards (starting around £9) and active cards, like Diva, Digium etc(~£300).Whilst the HFC cards work (with bristuff) you need to be prepared to reload modules regularly and go through other hoops.I used to work in a IT company, and there it's perfectly allright to usecheap cards, because the skills to reset modules etc are available atall times. For a clients' system I wouldn't go down that route and spend the money.I have no complains on call quality or dropped calls on cheap cards noron expensive cards, it's the administration and 'shinyness of theproduct'. Conrad___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2 * servers Host=ip - doesn't work Host=dynamic with register is OK, why?
Hi all,I've 2 * servers with static IP, and i notice that if i set both sip peers with host=server_ip and qualify=yes it presents UNREACHABLE on asterisk CLI.When i changed the host parameter to host=dynamic and set the register string in the [general] of sip.conf on both servers, the connection has been reestablished.I must say the 2 servers are in the same LAN with static IP.What could be the problem?-- Best regardsMarco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] announcing inbound PSTN calls
Hi,This is piece of cake for asterisk, but you need to do your script, or dialplan programing, asterisk has all the functions and applications to do it.But you need to get hands on it :) On 11/10/06, Jeronimo Romero [EMAIL PROTECTED] wrote: I'm running asterisk 1.2.8. I would like PSTN inbound calls to do the following: 1-once PSTN callers enter their desired extension; they have to record their name 2-recording then announces that it is trying to locate the user 3-asterisk calls local extension and announces callers recorded name 4-local recipient user can choose to take the call, send it to voicemail or transfer it to another extension Is this possible in asterisk?? . If it is possible, what is the name of this function? Is this documented anywhere? What is the best approach to doing this? Thanks in advance ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] announcing inbound PSTN calls
I would recommend you to record files with a uniqueid like var ${TIMESTAMP}[incoming]exten = s,1,Answer()exten = s,2,Backgroud(enter-ext)exten = _XXX,1,Playback(enter-name)exten = _XXX,2,Set(filename=${TIMESTAMP})
exten = _XXX,2,Record(/tmp/prompt${filename}:wav)exten = _XXX,3,Dial(zap/1/${EXTEN},A(${filename}.wav))On 11/10/06, bails
[EMAIL PROTECTED] wrote:exten = _XXX,3,Dial(zap/1/${EXTEN},,A(somefile))
bailsGustavo Berman wrote: Interesting! I think this can help for a start (but I don't know how to continue!!): [incoming] exten = s,1,Answer() exten = s,2,Backgroud(enter-ext)
exten = _XXX,1,Playback(enter-name) exten = _XXX,2,Record(/tmp/prompt${EXTEN}:wav) exten = _XXX,3,Dial(zap/1/${EXTEN}) now, how to play the recorded message to the called party when he/she
answers the phone? any help? On 11/10/06, * Jeronimo Romero* [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED] wrote: I'm running asterisk 1.2.8. I would like PSTN inbound calls to do the following: 1-once PSTN callers enter their desired extension; they have to
record their name 2-recording then announces that it is trying to locate the user 3-asterisk calls local extension and announces callers recorded name 4-local recipient user can choose to take the call, send it to
voicemail or transfer it to another extension Is this possible in asterisk?? . If it is possible, what is the name of this function? Is this documented anywhere?
What is the best approach to doing this? Thanks in advance ___
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Re: [asterisk-users] Alcatel trunk with asterisk problem on dialing digit-by-digit
Frederico,Pls Post your zapata.conf, any ways pls read bellow:On 11/9/06, Frederico Madeira [EMAIL PROTECTED]
wrote:Hi guys,I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P.
Input callsVOIP Proider --- Asterisk --- Alcatel
Output CallsVOIP Proider --- Asterisk --- AlcatelIn alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems:1. When users dial 2 on phone (alcatel) they don't received a dial tone, only receive a ocuped tone;
2. When users make step one, in asterisk console i received this message:!! Unexpected Channel selection 3
-- Extension '' in context 'default' from '027' does not exist. Rejecting call on channel 0/31, span 1If i configure in alcatel short dialing such: if user dial 3020 alcatel sent do asterisk a block number 31122332. In this case works fine.
How i can solve this problem ??Bellow i list my extension.conf[default]ignorepat=0
### Internal Calls## Input Calls exten= 312120XX,1,Dial(Zap/g1/${EXTEN:-4})
exten= 312120XX,2,HangupYou are missing _ for pattern match:exten= _312120XX,1,Dial(Zap/g1/${EXTEN
:-4})exten= _312120XX,2,Hangup
### External Callsexten= _,1,Dial(SIP/[EMAIL PROTECTED],60,Tt) # Local calls
exten= _,2,Hangupexten= _0XX,1,Dial(SIP/[EMAIL PROTECTED],60,Tt) # Long distance Callsexten= _0XX,2Hangupexten= _00XX,1,Dial(SIP/[EMAIL PROTECTED]
,60,Tt) # Internacional Calls
exten= _00XX,2HangupThanks.-- Frederico Madeira[EMAIL PROTECTED]
www.madeira.eng.br
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Re: [asterisk-users] Alcatel trunk with asterisk problem on dialing digit-by-digit
pls post it complete, i can't see there your channels for TE110P 30 voice channels...Also do this:[default]exten= _X.,1,Answer()exten= _X.,2,Noop(This is debug, i'm receive from Alcatel:${EXTEN})
exten= _X.,3,Wait()exten= _X.,4,Playback(vm-goodbye)exten= _X.,5,Hangupexten= h,1,hanguppls post the debug of incoming call from alcatel to * on you CLI .
On 11/9/06, Frederico Madeira [EMAIL PROTECTED] wrote:
Follow bellow:[trunkgroups][channels]language=ukcontext=defaultswitchtype=euroisdnsignalling=pri_netrxwink=300 usecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yes
callwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0group=1callgroup=1
pickupgroup=1immediate=noThanks Marco-- Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br2006/11/9, Marco Mouta
[EMAIL PROTECTED]:Frederico,
Pls Post your zapata.conf, any ways pls read bellow:
On 11/9/06, Frederico Madeira
[EMAIL PROTECTED]
wrote:Hi guys,I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P.
Input callsVOIP Proider --- Asterisk --- Alcatel
Output CallsVOIP Proider --- Asterisk --- AlcatelIn alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems:1. When users dial 2 on phone (alcatel) they don't received a dial tone, only receive a ocuped tone;
2. When users make step one, in asterisk console i received this message:!! Unexpected Channel selection 3
-- Extension '' in context 'default' from '027' does not exist. Rejecting call on channel 0/31, span 1If i configure in alcatel short dialing such: if user dial 3020 alcatel sent do asterisk a block number 31122332. In this case works fine.
How i can solve this problem ??Bellow i list my extension.conf[default]ignorepat=0
### Internal Calls## Input Calls exten= 312120XX,1,Dial(Zap/g1/${EXTEN:-4})
exten= 312120XX,2,HangupYou are missing _ for pattern match:exten= _312120XX,1,Dial(Zap/g1/${EXTEN
:-4})exten= _312120XX,2,Hangup
### External Callsexten= _,1,Dial(SIP/[EMAIL PROTECTED],60,Tt) # Local calls
exten= _,2,Hangupexten= _0XX,1,Dial(SIP/[EMAIL PROTECTED],60,Tt) # Long distance Callsexten= _0XX,2Hangupexten= _00XX,1,Dial(SIP/[EMAIL PROTECTED]
,60,Tt) # Internacional Calls
exten= _00XX,2HangupThanks.-- Frederico Madeira[EMAIL PROTECTED]
www.madeira.eng.br
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[EMAIL PROTECTED]
www.madeira.eng.br
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[asterisk-users] How do i redirect a call without answering it? SIP channel
Hi guys, I've been looking on wiki, but i could find it only for chan_capi: http://www.voip-info.org/wiki/view/Asterisk+PBX+functions In the CAPI channel See Asterisk CAPI channels * Call Deflection (CD) (redirect without answering): Implemented by chan_capi How can i do it with my Softphone Xlite? Any one can help me? I want to redirect a call without answering it. Best regards, MoutaPT ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diva server 4bri and Portuguese BRI
://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip realtime broken?
i've been using it sucessfully On 10/31/06, Don [EMAIL PROTECTED] wrote: Can someone confirm if sip realtime is broken in 1.2.13 and if so when was the last release it wasn't? heh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Server
Yes, you just need to setup asterisk with Digium board on the same server of your sipserver, and then you must establish a trunk between your sip server and asterisk. Then you must route calls using asterisk dialplan as well as your sip server dialplan. Be aware that if you have both on same server you must change SIP port in one of them. On 10/30/06, Imran M Yousuf [EMAIL PROTECTED] wrote: Hi Dear Users, I am new to Asterisk and had a query which is probably primitive. I wanted to know whether I can use the Digium Hardware and receive and establish connection to a host SIP Server which is totally a different platform. Let me explain - Usually there is a E1-VoIP gateway (independent Hardware) connecting to a Server/Client via LAN. In my case, SIP Server. Now what I want is that Digium PCI Hardware and the SIP Server will be the same PC and I Want the PCI Hardware to act as the gateway. Therefore my question in particular is: That is can I configure the device to talk to the Server in SIP protocol directly? -- Imran M Yousuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diva server 4bri and Portuguese BRI
pls post your misdn.conf as well as extensions.confMay be i can help.Sou Português:)On 10/29/06, Pedro Silva [EMAIL PROTECTED] wrote:Thanks Alberto!I tested with telsampl like you said (with various configurations for de diva) and this not works...:(The trace is:Enter destination address: 273xx--Conn_Req(273xx)Connect_Con--[29]:Disc_IndDisc_Res**Call cleared*** Any idea for the possible problem?Thanks and best regards,PS.2006/10/29, Alberto Pastore [EMAIL PROTECTED]: Pedro Silva ha scritto: Hello, I need to connect one diva server 4bri to a portuguese BRI interface. The operator (PT) said that this bri is in point-to-multipoint mode (S0). Previously one PBX has connected to that interface. The asterisk and diva drivers are working ok but i cannot communicate to outside via this bri. Xlite gives me the message: call failed: declined. Anyone have experience with this setup? What are the main parameters for bri card configuration? D-channel protocol: ETSI-DSS1 or other? Interface mode: NT or TE? Direct Inward Dialing (DID): Yes ou no? (MSN ou DID?) Thanks by any kind of help! Best regards, PS. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I'm not sure about Portuguese operators standard, but I bet ETSI-DSS1 should work just fine. The interface mode is surely TE. The DID/MSN should not affect outgoing calls, I generally leave DID off unless the telco company has that service active. If you're using the diva server for linux package from eicon (divas4linux, currently rel. 8.2), you should find a very simple utility named telsampl under /usr/lib/eicon/divas which you can run besides asterisk, to test outgoing calls. You should run it with this command line: telsampl -c x where x is the bri port you wish to test (1..4) then at the prompt type c and enter a pstn number, e.g. your mobile phone, then you can watch the log onscreen. If the outgoing call works, then your isdn setup is correct, and the problem is in asterisk. The message from xlite is not meaningful, as it could occur on many situations. You should watch the debug output on asterisk console. That helped me a lot. Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iax bug ?
pls post iax.conf of Both machines , as well as your dial() string on
both servers to connect each other.
That way would be easier to help you.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :
[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes
#include iax.voip1.conf
#include iax.renoir.conf
The iax.voip1.conf file contains :
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes
The iax.renoir.conf file contains :
[VOIP_RENOIR]
type=friend
host=renoir.lucyde
auth=rsa
inkey=key_184
outkey=key_Renoir
context=CONTEXT_RENOIR
trunk=yes
allow=gsm
Thanks to the variable context, when .184 receive a call from .160, this
call should be executed in the CONTEXT_VOIP1. In fact the call is
executed in the CONTEXT_RENOIR. Exactly (with a lot of test and debug),
the call is executed in the context of the last section's context of the
iax.conf file (e.g. CONTEXT_RENOIR here).
Anyone who has any idea ?
Thanks,
jb
PS :
(The debug in the .184 machine :
-- Accepting UNAUTHENTICATED call from 10.0.0.160:
requested format = ulaw,
requested prefs = (alaw),
actual format = gsm,
host prefs = (gsm),
priority = mine
-- Executing NoOp(IAX2/10.0.0.160:4569-1, I'm in
CONTEXT_RENOIR) in new stack
-- Executing Macro(IAX2/10.0.0.160:4569-1, check_forward|106)
in new stack
with the following extensions.conf :
[CONTEXT_VOIP1]
exten = _X.,1,NoOp(I'm in CONTEXT_VOIP1)
exten = _X.,2,Macro(check_forward,${EXTEN})
[CONTEXT_RENOIR]
exten = _X.,1,NoOp(I'm in CONTEXT_RENOIR)
exten = _X.,2,Macro(check_forward,${EXTEN})
)
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--
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Marco Mouta
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Re: [asterisk-users] Iax bug ?
Hi,
Unfortunately i'm not able to debug this with you now :( I'm busy.
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
This secret empty is this allowed?
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes
Try a simple test with this, and then step by step go to rsa authentication.
http://astrecipes.net/index.php?n=204
If in troubles, post here i'll try to help you
By the way, to understand much better what's going on i would
recommend you to not use type=friend and use type=user and type=peer.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Here the .160's iax.conf file :
[general]
bandwidth=high
tos=reliability
bandwidth=low
disallow=all; Icky sound quality... Mr. Roboto.
allow=alaw ; Always allow GSM, it's cool :)
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
[VOIP1]
type=friend
host=10.0.0.184
auth=rsa
inkey=voip3
outkey=voip1
context=VOIPLINK3
qualify=1
trunk=yes
allow=all
How .160 call .184 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
How .184 call .160 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
(the same)
Thanks,
jb
Marco Mouta a écrit :
pls post iax.conf of Both machines , as well as your dial() string on
both servers to connect each other.
That way would be easier to help you.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :
[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes
#include iax.voip1.conf
#include iax.renoir.conf
The iax.voip1.conf file contains :
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes
The iax.renoir.conf file contains :
[VOIP_RENOIR]
type=friend
host=renoir.lucyde
auth=rsa
inkey=key_184
outkey=key_Renoir
context=CONTEXT_RENOIR
trunk=yes
allow=gsm
Thanks to the variable context, when .184 receive a call from .160, this
call should be executed in the CONTEXT_VOIP1. In fact the call is
executed in the CONTEXT_RENOIR. Exactly (with a lot of test and debug),
the call is executed in the context of the last section's context of the
iax.conf file (e.g. CONTEXT_RENOIR here).
Anyone who has any idea ?
Thanks,
jb
PS :
(The debug in the .184 machine :
-- Accepting UNAUTHENTICATED call from 10.0.0.160:
requested format = ulaw,
requested prefs = (alaw),
actual format = gsm,
host prefs = (gsm),
priority = mine
-- Executing NoOp(IAX2/10.0.0.160:4569-1, I'm in
CONTEXT_RENOIR) in new stack
-- Executing Macro(IAX2/10.0.0.160:4569-1, check_forward|106)
in new stack
with the following extensions.conf :
[CONTEXT_VOIP1]
exten = _X.,1,NoOp(I'm in CONTEXT_VOIP1)
exten = _X.,2,Macro(check_forward,${EXTEN})
[CONTEXT_RENOIR]
exten = _X.,1,NoOp(I'm in CONTEXT_RENOIR)
exten = _X.,2,Macro(check_forward,${EXTEN})
)
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--
Jean-Baptiste Bellet
Ingénieur Développpement
Lucyde SAS
Prologue 1 - La Pyrénéenne BP 27201 LABEGE cedex
+33 (0)5 34 31 86 36
http://www.lucyde.com
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--
Com os melhores cumprimentos,
Marco Mouta
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Re: [asterisk-users] Iax bug ?
Hi,
I think i found your problem, look that in your debug you have, -
Accepting UNAUTHENTICATED call from 10.0.0.160:
Take a look on incoming call authentication, and how asterisk handles this:
http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication
Incoming Connections
When Asterisk receives an incoming IAX connection, the initial call
information can include a username (in the IAX2 USERNAME field) or
not. In addition, the incoming connection has a source IP address that
Asterisk can use for authentication as well.
If a username is supplied, Asterisk does the following:
* Search iax.conf for a type=user entry with a section name (eg
[username]) matching the supplied username; if no matching entry is
found, refuse the connection.
* If the found entry has allow and/or deny settings, compare the
IP address of the caller to these lists. If the connection is not
allowed, refuse the connection.
* Perform the desired secret checking (plaintext, md5 or rsa); if
it fails, refuse the connection.
* Accept the connection and send the caller to the context
specified in the context setting for this iax.conf entry.
If a username is not supplied, Asterisk does the following:
* Search for a type=user entry in iax.conf with no secret
specified and also allow and/or deny restrictions that do not restrict
the caller from connecting. If such an entry is found, accept the
connection, and use the name of the found iax.conf entry as the
connecting username.
* Search for a type=user entry in iax.conf with no secret
specified and no allow and/or deny restrictions at all. If such an
entry is found, accept the connection. and use the name of the found
iax.conf entry as the connecting username.
* Search for a type=user entry in iax.conf with a secret (or RSA
key) specified and also allow and/or deny restrictions that do not
restrict the caller from connecting. If such an entry is found,
attempt to authenticate the caller using the specified secret or key,
and if that passes, accept the connection, and use the name of the
found iax.conf entry as the connecting username.
* Search for a type=user entry in iax.conf with a secret (or RSA
key) specified and no allow and/or deny restrictions at all. If such
an entry is found, attempt to authenticate the caller using the
specified secret or key, and if that passes, accept the connection,
and use the name of the found iax.conf entry as the connecting
username.
Hope this helps!
I didn't read all, but what i guess is: the incoming call isn't being
correctly authenticated, so can't go to VOIP1 as you desire, then as
is mention above:
Search for a type=user entry in iax.conf with no secret specified
and no allow and/or deny restrictions at all. If such an entry is
found, accept the connection. and use the name of the found iax.conf
entry as the connecting username.
Pls give some feedback if you solved the problem.
On 10/27/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,
Unfortunately i'm not able to debug this with you now :( I'm busy.
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
This secret empty is this allowed?
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes
Try a simple test with this, and then step by step go to rsa authentication.
http://astrecipes.net/index.php?n=204
If in troubles, post here i'll try to help you
By the way, to understand much better what's going on i would
recommend you to not use type=friend and use type=user and type=peer.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Here the .160's iax.conf file :
[general]
bandwidth=high
tos=reliability
bandwidth=low
disallow=all; Icky sound quality... Mr. Roboto.
allow=alaw ; Always allow GSM, it's cool :)
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
[VOIP1]
type=friend
host=10.0.0.184
auth=rsa
inkey=voip3
outkey=voip1
context=VOIPLINK3
qualify=1
trunk=yes
allow=all
How .160 call .184 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
How .184 call .160 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
(the same)
Thanks,
jb
Marco Mouta a écrit :
pls post iax.conf of Both machines , as well as your dial() string on
both servers to connect each other.
That way would be easier to help you.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :
[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes
#include iax.voip1.conf
#include iax.renoir.conf
The iax.voip1.conf file contains :
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk
Re: [asterisk-users] Direct call vs Block call
pls post your misdn.conf as well as extensions.conf, so someone could help you on this. On 10/27/06, Frederico Madeira [EMAIL PROTECTED] wrote: Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. For alcatel users use asterisk lines, should dial 0 to take tone from asterisk. In default configuration in alcatel, if user dial 0 this error occour: !! Unexpected Channel selection 3 -- Extension '' in context 'default' from '' does not exist. Rejecting call on channel 0/31, span 1 In alcatel we're enable block dial, so alcatel only send to asterisk when user end dialing all digits. How i permit the first case to work ?? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk misdn incoming line not working.
Plse Read bellow:
On 10/27/06, Mark Hannessen [EMAIL PROTECTED] wrote:
Hi list,
I have a server running a simple hfs isdn card running with chan_misdn.
the problem is, I can't get asterisk to pick up the phone, outgoing calls work
fine.
when running asterisk with -vvvc I get the following log when I try
to dial the isdn server.
P[ 1] * Starting Ast ctx:kpn-in dad:0594643637 oad:0878722291 with 's'
extension
P[ 1] MGMT: SSTATUS: L1_ACTIVATED
== Starting mISDN/1-2 at kpn-in,s,1 failed so falling back to exten 's'
== Starting mISDN/1-2 at kpn-in,s,1 still failed so falling back to
context 'default'
Oct 27 01:28:42 WARNING[3514]: pbx.c:2357 __ast_pbx_run: Channel 'mISDN/1-2'
sent into invalid extension 's' in context 'default', but no invalid handler
087822291 is the number i dial from, 0594643637 is the number that the
asterisk server should respond to.
in misdn.conf i created a kpn section like this:
[kpn]
ports=1ptmp
;group=1
immediate=yes
always_immediate=yes
context=kpn-in
hold_allowed=yes
msns=*
and in extensions.conf i created a very basic kpn-in section like this:
[kpn-in]
HERE IS your PROBLEM:
exten = kpn-in,1,Dial(SIP/mark,25,tr)
1- Be sure of of MSNs string your telco is sending you.
2- Do this:
[kpn-is]
exten= _X.,1,answer
exten= _X.,1,Noop(My telco is sending me this MSN string: ${EXTEN})
exten= _X.,3,wait(1)
exten= _X.,4,Playback(vm-goodbye)
exten= _X.,5,hangup
reload your asterisk after this changes, and dial again.
Now you may understand what your telco is sending you and then start
routing it on your way.
Hope this helps,
Pls. give me some feedback.
I don't really have much experience with asterisk so I probably did something
wrong here, but I couldn't really figure out how to get it done.
anyone out there any ideas?
Mark
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Marco Mouta
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Re: [asterisk-users] asterisk misdn incoming line not working.
My mistake:
[kpn-is]
exten= _X.,1,answer
exten= _X.,2,Noop(My telco is sending me this MSN string: ${EXTEN})
exten= _X.,3,wait(1)
exten= _X.,4,Playback(vm-goodbye)
exten= _X.,5,hangup
On 10/27/06, Marco Mouta [EMAIL PROTECTED] wrote:
Plse Read bellow:
On 10/27/06, Mark Hannessen [EMAIL PROTECTED] wrote:
Hi list,
I have a server running a simple hfs isdn card running with chan_misdn.
the problem is, I can't get asterisk to pick up the phone, outgoing calls work
fine.
when running asterisk with -vvvc I get the following log when I try
to dial the isdn server.
P[ 1] * Starting Ast ctx:kpn-in dad:0594643637 oad:0878722291 with 's'
extension
P[ 1] MGMT: SSTATUS: L1_ACTIVATED
== Starting mISDN/1-2 at kpn-in,s,1 failed so falling back to exten 's'
== Starting mISDN/1-2 at kpn-in,s,1 still failed so falling back to
context 'default'
Oct 27 01:28:42 WARNING[3514]: pbx.c:2357 __ast_pbx_run: Channel 'mISDN/1-2'
sent into invalid extension 's' in context 'default', but no invalid handler
087822291 is the number i dial from, 0594643637 is the number that the
asterisk server should respond to.
in misdn.conf i created a kpn section like this:
[kpn]
ports=1ptmp
;group=1
immediate=yes
always_immediate=yes
context=kpn-in
hold_allowed=yes
msns=*
and in extensions.conf i created a very basic kpn-in section like this:
[kpn-in]
HERE IS your PROBLEM:
exten = kpn-in,1,Dial(SIP/mark,25,tr)
1- Be sure of of MSNs string your telco is sending you.
2- Do this:
[kpn-is]
exten= _X.,1,answer
exten= _X.,1,Noop(My telco is sending me this MSN string: ${EXTEN})
exten= _X.,3,wait(1)
exten= _X.,4,Playback(vm-goodbye)
exten= _X.,5,hangup
reload your asterisk after this changes, and dial again.
Now you may understand what your telco is sending you and then start
routing it on your way.
Hope this helps,
Pls. give me some feedback.
I don't really have much experience with asterisk so I probably did something
wrong here, but I couldn't really figure out how to get it done.
anyone out there any ideas?
Mark
___
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asterisk-users mailing list
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--
Com os melhores cumprimentos,
Marco Mouta
--
Com os melhores cumprimentos,
Marco Mouta
___
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Re: [asterisk-users] Iax bug ?
Why r u using rsa authentication? you should start with something
simple. test the link i sent u.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Thanks a lot.
I think UNAUTHENTICATED call is the source of my problems.
How I can solve it ?
Because allowguest is a sip.conf option ...
jb
Marco Mouta a écrit :
Hi,
I think i found your problem, look that in your debug you have, -
Accepting UNAUTHENTICATED call from 10.0.0.160:
Take a look on incoming call authentication, and how asterisk handles this:
http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication
Incoming Connections
When Asterisk receives an incoming IAX connection, the initial call
information can include a username (in the IAX2 USERNAME field) or
not. In addition, the incoming connection has a source IP address that
Asterisk can use for authentication as well.
If a username is supplied, Asterisk does the following:
* Search iax.conf for a type=user entry with a section name (eg
[username]) matching the supplied username; if no matching entry is
found, refuse the connection.
* If the found entry has allow and/or deny settings, compare the
IP address of the caller to these lists. If the connection is not
allowed, refuse the connection.
* Perform the desired secret checking (plaintext, md5 or rsa); if
it fails, refuse the connection.
* Accept the connection and send the caller to the context
specified in the context setting for this iax.conf entry.
If a username is not supplied, Asterisk does the following:
* Search for a type=user entry in iax.conf with no secret
specified and also allow and/or deny restrictions that do not restrict
the caller from connecting. If such an entry is found, accept the
connection, and use the name of the found iax.conf entry as the
connecting username.
* Search for a type=user entry in iax.conf with no secret
specified and no allow and/or deny restrictions at all. If such an
entry is found, accept the connection. and use the name of the found
iax.conf entry as the connecting username.
* Search for a type=user entry in iax.conf with a secret (or RSA
key) specified and also allow and/or deny restrictions that do not
restrict the caller from connecting. If such an entry is found,
attempt to authenticate the caller using the specified secret or key,
and if that passes, accept the connection, and use the name of the
found iax.conf entry as the connecting username.
* Search for a type=user entry in iax.conf with a secret (or RSA
key) specified and no allow and/or deny restrictions at all. If such
an entry is found, attempt to authenticate the caller using the
specified secret or key, and if that passes, accept the connection,
and use the name of the found iax.conf entry as the connecting
username.
Hope this helps!
I didn't read all, but what i guess is: the incoming call isn't being
correctly authenticated, so can't go to VOIP1 as you desire, then as
is mention above:
Search for a type=user entry in iax.conf with no secret specified
and no allow and/or deny restrictions at all. If such an entry is
found, accept the connection. and use the name of the found iax.conf
entry as the connecting username.
Pls give some feedback if you solved the problem.
On 10/27/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,
Unfortunately i'm not able to debug this with you now :( I'm busy.
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
This secret empty is this allowed?
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes
Try a simple test with this, and then step by step go to rsa
authentication.
http://astrecipes.net/index.php?n=204
If in troubles, post here i'll try to help you
By the way, to understand much better what's going on i would
recommend you to not use type=friend and use type=user and type=peer.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Here the .160's iax.conf file :
[general]
bandwidth=high
tos=reliability
bandwidth=low
disallow=all; Icky sound quality... Mr. Roboto.
allow=alaw ; Always allow GSM, it's cool :)
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
[VOIP1]
type=friend
host=10.0.0.184
auth=rsa
inkey=voip3
outkey=voip1
context=VOIPLINK3
qualify=1
trunk=yes
allow=all
How .160 call .184 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
How .184 call .160 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
(the same)
Thanks,
jb
Marco Mouta a écrit :
pls post iax.conf of Both machines , as well as your dial() string on
both servers to connect each other.
That way would be easier to help you.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot
