[asterisk-users] Asterisk with app_RPT question
Dear All, I am not sure if this is the right place to ask my question but I can't find a newsgroup or support for this app_RPT concept so I hope if some one in this community who have tried it out could help me out. I studied this application requirments and saw the hardware needed they describe a radio quad which uses RJ 45 but I can't see where the RJ goes in order to be able to communicate with the radio devices, I have absulotly no knowledge in two way radio devices so I hope some one could complete the full picture for me. And what I plan to use this feauture is to expand my * PBX ntwork to remote sites which have the radio quad and use it to talk to onsite engineers through the two way radio devices (such as motorola) using iaxrpt. -- Thx MAG ___ --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] groups
Dear Khaled, The way I would go to do so is to put the group of people you want to call each other in one context and the other people in an another context. That's one way to do so. Thx MAG Khaled wrote: Dears Please how can create an independent group of users on asterisk ,in which user on group A cant dial user on group B. Thanks Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 --- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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 -- Thx MAG ___ --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] Cisco 7960
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Re: [asterisk-users] Cisco 7960
Dear Khaled, What is the softphone u r using? Thx MAG Khaled wrote: I am using firmware version pos3-07-500 Kindly can you provide me with the basic configuration for cisco ip phone and asterisk config file *I have nat=never at my asterisk config file and nat enabled N0 at cisco phone *I have an out bound proxy ip and port 5060 at cisco phone *Voip control port is 5061 My problem is my soft phone can call the cisco phone with normal RTP and Bye message,but my cisco phone cant dial my soft phone. Asterisk sends bye message for my soft phone. Thanks --- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wireless Sent: Tuesday, February 27, 2007 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 can you give a bit more info? I know that you need nat=never for example - Original Message - From:Khaled To:'Asterisk Users Mailing List - Non-Commercial Discussion' Cc:[EMAIL PROTECTED] Sent: Tuesday, February 27, 2007 10:03 AM Subject: [asterisk-users] Cisco 7960 Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 - * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. - ___ --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 --- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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 -- Thx MAG ___ --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: FW: [asterisk-users] Cisco 7960
Dear All, Please send the sip configuration for both phones along with a debug from asterisk when you try to call from cisco to the eyebeam? also are you trying to make them call peer to peer or not? What I am suspecting is that there must be something mismatching when the cisco phone tries to call the softphone you just need to focus on the debug and check the configuration. Thx MAG Khaled wrote: Softphone Eyebeam v 1.5.2 --- From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohamed A. Gombolaty Sent: Tuesday, February 27, 2007 2:03 PM To:Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 Dear Khaled, What is the softphone u r using? Thx MAG Khaled wrote: I am using firmware version pos3-07-500 Kindly can you provide me with the basic configuration for cisco ip phone and asterisk config file *I have nat=never at my asterisk config file and nat enabled N0 at cisco phone *I have an out bound proxy ip and port 5060 at cisco phone *Voip control port is 5061 My problem is my soft phone can call the cisco phone with normal RTP and Bye message,but my cisco phone cant dial my soft phone. Asterisk sends bye message for my soft phone. Thanks - From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wireless Sent: Tuesday, February 27, 2007 12:48 PM To:Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 can you give a bit more info? I know that you need nat=never for example - Original Message - From:Khaled To:'Asterisk Users Mailing List - Non-Commercial Discussion' Cc:[EMAIL PROTECTED] Sent: Tuesday, February 27, 2007 10:03 AM Subject: [asterisk-users] Cisco 7960 Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 --- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. --- ___ --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 - * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity
Re: [asterisk-users] Do I understand GROUPs correctly?
Dear Mike,
I had wanted to do something that is similar to your need as I wanted to be
able to add one active channel in multiple groups, it worked with The Ramon's
example in the link below which uses categories beside the set command, note
there are two examles depending on the asterisk version you are using:
http://www.voip-info.org/wiki/view/asterisk+cmd+setgroup
Thx
MAG
Mike wrote:
Ok, that sort of makes sense. But what I am doing is passing off a call
into my Asterisk system to a cell phone. I want this to count as 2
channels. So, I am doing, in effect, this kind of algo:
Answer the call
Set(Group) to increment channel to 1
Play IVR, go into menus, etc.
Eventually go into a Set(group) again to increment channel before dialing a
cell phone using a dial(cellphone#) cmd.
If that doesn't work, how do I accomplish the same kind of thing elegantly?
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Kempgen
Sent: Tuesday, February 27, 2007 10:57
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Do I understand GROUPs correctly?
Doug Lytle wrote:
Mike wrote:
Hi,
I was under the impression that Set(GROUP()=1234) incremented some
value associated with 1234.
So if I did the same thing twice, I'd get a group count of 2.
Ex:
exten = s,1,Set(GROUP()=1234)
exten = s,n,Set(GROUP()=1234)
exten = s,n,Noop(Used channels: ${GROUP_COUNT(1234})
If this is a direct copy/paste then your error is in line 3. You have
a } positioned incorrectly. My example below:
exten = _35XX,1,Set(GROUP()=Max_Calls) exten = _35XX,n,NoOP(Active
Calls: ${GROUP_COUNT(Max_Calls)})
Apart from that you assign the group 1234 twice to the *same* channel. So
GROUP_COUNT(1234) correctly reports only *1* channel to be in that group.
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
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Thx
MAG
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[asterisk-users] Calls die when the answering party transfers
Dear All, I am facing a strange problem that I can't find any matches for while googling, sometimes while a call initiated from asterisk to the PSTN is answered and the answering party say the receiptionist tries to transfer the call to someone else, the call dies, the full log shows nothing useful and I am really unable to move forward on this issue, so can some one suggest anything? My zapata.conf is below also we are using Digium TDM400P with FXO modules to connect to the PSTN. [channels] callerid = asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 busydetect=yes immediate=no faxdetect=both busycount=4 callgroup=1 pickupgroup=1 pridialplan = local prilocaldialplan = local nationalprefix = 1 internationalprefix = 1011 group = 0 context=from-pstn signalling=pri_cpe switchtype = euroisdn language=en channel = 1-15,17-31 signalling=fxs_ks context=from-zaptel group=3 channel = 63-74 signalling=fxs_ks context=from-zaptel group=4 channel = 75-78 -- Thx MAG ___ --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] Fax killed on all zaptel devices
Dear All, I have this problem which is preventing me from switching to voip system andstill working on that old siemens pbx, we have fax machines that we attached to ATA called planet and when we try to send a fax locally between the fax machines it works great but when we try to get a fax machine to send or recieve on the E1 pri or on a TDM400p (notice all cards are digium) it get's a communication error, here is a step by step of what happens wjen we try to send a fax from the outside to asterisk (this problem happens to both fax machines and the making asterisk recieving the fax and send it by mail): - the phone rings and the fax picks up the call - the fax starts to do it's sound but it seems to get cut just after one second - I get on the cli of asterisk: unknown rtp codecs 109 recieved - then the fax after say a minut or so on the sending side gives a communication error and closes. Does anyone has an idea I really need it to work either on the fax machines or the mail option available. -- Thx MAG ___ --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] Re: Outgoing problem on PRI
Dear All,
The resolution to the problem below was very easy and I guess that what
made it very hard:
callerid=asreceived
signalling=pri_cpe
switchtype=> euroisdn
context=from-zaptel
group=0
channel=>1-15,17-31
Thx
MAG
"Mohamed A. Gombolaty" wrote:
Dear All,
I have an asterisk server version 1.2.12.1 along with trixbox
and I am having this nasty problem, I have a TE200P and have an E1 pri
attached to it and zttool says it's OK, I have configured the whole
31 channels into one group as follow:
Zapata-auto.conf:
callerid=asreceived
signalling=pri_cpe
switchtype=euroisdn
context=from-zaptel
group=0
channel=>1-15,17-31
/etc/zaptel.conf:
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
Now I can recieve calls on the pri and everyhting is well but I can't
make calls from the pri, whenever I try I get all circuits are busy message
here is a log from asterisk cli when I try to make a call out using pri
it is a tiny long but trixbox does add many macros and stuff put
I do have suspicions about what can cause the zap channel to get a Hungup
request as it seems from below that is the case :
-- Executing Macro("SIP/146-b78060b0", "dialout-trunk|3|6536595||")in
new stack
-- Executing GotoIf("SIP/146-b78060b0", "1?3:2")
in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/146-b78060b0", "user-callerid")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?report")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?start")
in new stack
-- Executing Set("SIP/146-b78060b0", "REALCALLERIDNUM=146")
in newstack
-- Executing NoOp("SIP/146-b78060b0", "REALCALLERIDNUM
is 146") in new stack
-- Executing Set("SIP/146-b78060b0", "AMPUSER=146")
in new stack
-- Executing Set("SIP/146-b78060b0", "AMPUSERCIDNAME=Mohamed
Samir -UNIX") in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?report")
in new stack
-- Executing Set("SIP/146-b78060b0", "CALLERID(all)=Mohamed
Samir - UNIX 146>") in new stack
-- Executing NoOp("SIP/146-b78060b0", "Using CallerID
"Mohamed Samir - UNIX" 146>") in new stack
-- Executing Macro("SIP/146-b78060b0", "record-enable|146|OUT")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0 > 0?2:4")
in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/146-b78060b0", "recordingcheck|20061110-162404|1163168644.20")
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
-- Executing Macro("SIP/146-b78060b0", "dialout-trunk|3|6536595||")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "1?3:2")
in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/146-b78060b0", "user-callerid")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?report")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?start")
in new stack
-- Executing Set("SIP/146-b78060b0", "REALCALLERIDNUM=146")
in new stack
-- Executing NoOp("SIP/146-b78060b0", "REALCALLERIDNUM
is 146") in new stack
-- Executing Set("SIP/146-b78060b0", "AMPUSER=146")
in new stack
-- Executing Set("SIP/146-b78060b0", "AMPUSERCIDNAME=Mohamed
Samir - UNIX") in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?report")
in new stack
-- Executing Set("SIP/146-b78060b0", "CALLERID(all)=Mohamed
Samir - UNIX 146>") in new stack
-- Executing NoOp("SIP/146-b78060b0", "Using CallerID
"Mohamed Samir - UNIX" 146>") in new stack
-- Executing Macro("SIP/146-b78060b0", "record-enable|146|OUT")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0 > 0?2:4")
in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/146-b78060b0", "recordingcheck|20061110-162404|1163168644.20")
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheckrecordingcheck|20061110-162404|1163168644.20:
Outbound recording not enabled
recordingcheck|20061110-162404|1163168644.20: Outbound recording
not enabled
-- AGI Script recordingcheck completed, returning
0
-- AGI Script recordingcheck completed, returning
0
-- Executing NoOp("SIP/146-b78060b0", "No recording
needed") in new stack
-- Executing NoOp("SIP/146-b78060b0", "No recording
needed") in new stack
-- Executing Macro("SIP/146-b78060b0", "outbound-callerid|3")
in new stack
-- Executing Macro("SIP/146-b78060b0", "outbound-callerid|3")
in new stack
-- Executin
[asterisk-users] Outgoing problem on PRI
Dear All,
I have an asterisk server version 1.2.12.1 along with trixbox
and I am having this nasty problem, I have a TE200P and have an E1 pri
attached to it and zttool says it's OK, I have configured the whole
31 channels into one group as follow:
Zapata-auto.conf:
callerid=asreceived
signalling=pri_cpe
switchtype=euroisdn
context=from-zaptel
group=0
channel=>1-15,17-31
/etc/zaptel.conf:
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
Now I can recieve calls on the pri and everyhting is well but I can't
make calls from the pri, whenever I try I get all circuits are busy message
here is a log from asterisk cli when I try to make a call out using pri
it is a tiny long but trixbox does add many macros and stuff put
I do have suspicions about what can cause the zap channel to get a Hungup
request as it seems from below that is the case :
-- Executing Macro("SIP/146-b78060b0", "dialout-trunk|3|6536595||")in
new stack
-- Executing GotoIf("SIP/146-b78060b0", "1?3:2")
in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/146-b78060b0", "user-callerid")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?report")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?start")
in new stack
-- Executing Set("SIP/146-b78060b0", "REALCALLERIDNUM=146")
in newstack
-- Executing NoOp("SIP/146-b78060b0", "REALCALLERIDNUM
is 146") in new stack
-- Executing Set("SIP/146-b78060b0", "AMPUSER=146")
in new stack
-- Executing Set("SIP/146-b78060b0", "AMPUSERCIDNAME=Mohamed
Samir -UNIX") in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?report")
in new stack
-- Executing Set("SIP/146-b78060b0", "CALLERID(all)=Mohamed
Samir - UNIX 146>") in new stack
-- Executing NoOp("SIP/146-b78060b0", "Using CallerID
"Mohamed Samir - UNIX" 146>") in new stack
-- Executing Macro("SIP/146-b78060b0", "record-enable|146|OUT")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0 > 0?2:4")
in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/146-b78060b0", "recordingcheck|20061110-162404|1163168644.20")
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
-- Executing Macro("SIP/146-b78060b0", "dialout-trunk|3|6536595||")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "1?3:2")
in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/146-b78060b0", "user-callerid")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?report")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?start")
in new stack
-- Executing Set("SIP/146-b78060b0", "REALCALLERIDNUM=146")
in new stack
-- Executing NoOp("SIP/146-b78060b0", "REALCALLERIDNUM
is 146") in new stack
-- Executing Set("SIP/146-b78060b0", "AMPUSER=146")
in new stack
-- Executing Set("SIP/146-b78060b0", "AMPUSERCIDNAME=Mohamed
Samir - UNIX") in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?report")
in new stack
-- Executing Set("SIP/146-b78060b0", "CALLERID(all)=Mohamed
Samir - UNIX 146>") in new stack
-- Executing NoOp("SIP/146-b78060b0", "Using CallerID
"Mohamed Samir - UNIX" 146>") in new stack
-- Executing Macro("SIP/146-b78060b0", "record-enable|146|OUT")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0 > 0?2:4")
in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/146-b78060b0", "recordingcheck|20061110-162404|1163168644.20")
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheckrecordingcheck|20061110-162404|1163168644.20:
Outbound recording not enabled
recordingcheck|20061110-162404|1163168644.20: Outbound recording
not enabled
-- AGI Script recordingcheck completed, returning
0
-- AGI Script recordingcheck completed, returning
0
-- Executing NoOp("SIP/146-b78060b0", "No recording
needed") in new stack
-- Executing NoOp("SIP/146-b78060b0", "No recording
needed") in new stack
-- Executing Macro("SIP/146-b78060b0", "outbound-callerid|3")
in new stack
-- Executing Macro("SIP/146-b78060b0", "outbound-callerid|3")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "1?start")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "1?start")
in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/146-b78060b0", "REALCALLERIDNUM
is 146") in new stack
-- Executing NoOp("SIP/146-b78060b0", "REALCALLERIDNUM
is 146") in new stack
-- Executing Set("SIP/146-b78060b0", "USEROUTCID=")
in new stack
-- Executing Set("SIP/146-b78060b0", "USEROUTCID=")
in new stack
-- Executing Set("SIP/146-b78060b0", "EMERGENCYCID=")
in new stack
-- Executing Set("SIP/146-b78060b0", "EMERGENCYCID=")
in new stack
-- Executing Set("SIP/146-b78060b0", "TRUNKOUTCID=")
in new stack
-- Executing Set("SIP/146-b78060b0", "TRUNKOUTCID=")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "1?trunkcid")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "1?trunkcid")
in new stack
-- Goto (macro-outbound-callerid,s,11)
-- Goto (macro-outbound-callerid,s,11)
-- Executing
Re: [asterisk-users] Compiling Zaptel 1.2.10 on Ubuntu 6.10
Dear Storm, I have two guesses One could be something in the ubuntu make which makes it unable to understand some regx in the scripts used or I am not quite sure but check the kernel version you are having (i do that by uname -a ) I believe you will find something there, if it is not the same as the one in ubuntu 6.06 then try installing the kernel of 6.06 else I have no idea. Thx' MAG Strom Carlson wrote: Here's a weird problem that I'm not quite sure how to resolve. Zaptel 1.2.10 compiles just fine with "make", but when "make install" is run, this happens: [ `id -u` = 0 ] /sbin/depmod -a 2.6.17-10-generic || : [ -f /etc/zaptel.conf ] || install -D -m 644 zaptel.conf.sample /etc/zaptel.conf build_tools/genmodconf linux26 "" "tor2 torisa wcusb wcfxo wctdm wctdm24xxp ztdynamic ztd-eth wct1xxp wcte11xp pciradio ztd-loc ztdummy" [: 66: ==: unexpected operator [: 66: ==: unexpected operator Unknown kernel build version requested... exiting. make: *** [install] Error 1 This worked just fine under ubuntu 6.06 with the same set of packages installed. Any help is appreciated. -- Strom Carlson http://www.stromcarlson.com/ ___ --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 -- Thx MAG ___ --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] Stopping putgoing calls after working hours
Thanx Jacob,
I did notice the locking phones at night mails in the list, and I have
just finished making the solution and it is just what I wanted here is
my addition in the extension.conf (with trixbox I did it in extensions_trixbox.conf),
note I have used the Authenticate command also with the astdb just serach
voip-info for the command and you will understand the drill, for each phone
number I added the password needed using the database put command
from console :
;MAG Addition for phone locking
;to lock extensions
exten => *00,1,Wait(1)
exten => *00,2,Answer
exten => *00,3,Authenticate(/${CALLERID(num)}|d) ; the caller will
be prompted for password
exten => *00,4,Set(DB(LOCKPHONE/${CALLERID(num)})=1) ; if passed
the lock variable will be set to equal one
exten => *00,5,Hangup
;to unlock extensions
exten => *01,1,Wait(1)
exten => *01,2,Answer
exten => *01,3,Authenticate(/${CALLERID(name)}|d) ; same as above
exten => *01,4,Set(DB(LOCKPHONE/${CALLERID(num)})=0); reverse of the
above
exten => *01,5,Hangup
All that is left is to make a line that checks the variable in astdb
when the calls are going to trunks and ofcourse made the trunk responsible
for emergency calls without this feature so you can only call the police,
ambulance, power and fire departments and internal extensions but not the
costly outside calls
Thx
MAG
Benjamin Jacob wrote:
Mohamed A. Gombolaty wrote:
> Dear Rich,
>
> It seems that my question is very general I apologize for that, but
I
> am glad to see others like yourself pointing me in different
> directions, it seems all around the world we have problems with the
> cleaning folks.
>
> What I have in mind is to make the phone user lock his phone when
he
> is leaving with a special code and relock it back when he comes to
> work (and
>
u mean unlock it..
> as for emergency calls there are attendants who work at night who
will
> be able to make an emergency call whenever needed at the spot), now
> there is nothing that seems to be able to do that directly, I have
> played around with the gotoiftime and also the time based dial plan
> include sent in mails before that.
>
> But while working I thought of another approach why not create a
php
> web interface that each user logs in with a special username and
> password and gives him access to lock his phone, and what php does
is
> actually change the secret password to something else than the
> configured on the phone, this should make the phone unable to
> authenticate thus not being able to make a call, and unlocking it
> returns the password to it's right form, I have already found the
> tables that I need to play around so I will restart making the php.
I
> will update the list back with my final result.
>
>
> Do you guys think I could send a mail to the dev site to see if they
> can add this feature to asterisk.
>
Am writing a few dialplans that you could use. I havent testted it..
u
might have to refine it.. am writing all this at runtime :-)
To lock and unlock phones, you need not go to php and change passwords
etc. You can use DB operations.
To lock phones, users can call into one particular number, e.g. *01
[lockphone]
exten => *01,1,Set(DB(LOCKPHONE/${CALLERID(num)})=1})
To unlock phones, u set the DB custom variable LOCKPHONE to zero,
using
another number, say *02
[unlockphone]
exten => *02,1,Set(DB(LOCKPHONE/${CALLERID(num)})=0})
So, to avoid calls, you'll have to check the value of this custom
variable everytime. To avoid repeated checks even in the day time,
you
can put the following dialplan, only in contexts which are invoked
at
night(read the previous posts).
[night-context]
exten => 911,1,Dial(Zap/999) ;;;wotever syntax, I've
never
worked with ZAP, for 911 emergency calls even at night.
include => lockphone
include => unlockphone
include => othernumbers
[othernumbers]
exten => _[0-9].,1, Set(locked=DB(LOCKPHONE/${CALLERID(num)}))
exten => _[0-9].,2,GotoIf($[${locked}=0]?:5)
allow call only
if phone is unlocked
exten => _[0-9].,3, Dial(SIP/${EXTEN})
phone is
unlocked , so call away to glory
exten => _[0-9].,4, Hangup
exten => _[0-9].,5, Playback(hussh-sleep-now) ;;; cant
call
now, cuz phones locked
exten => _[0-9].,6, Hangup
Now you lock n unlock ur phones whenever u want.
cheerz
- Ben.
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Thx
MAG
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Re: [asterisk-users] Stopping putgoing calls after working hours
Dear Lacy Thx Lacy for this important reminder we engineers do tend sometimes to forget about all the law part, indeed while I was putting down the implementation we do have exceptions we have a 24x7 call center and ofcourse the emergency number. Thx MAG Lacy Moore - Aspendora wrote: So I was wondering is there a way to make this happen in asterisk?? Depending on where you are located, you might want to allow emergency calls to go through. The bloodsuckers, I mean attorneys, here in the US would have a field day if something were to happen to someone at a company that did not allow emergency numbers to be dialed. Translated: If something were to happen to someone outside of business hours (in the US), and the phones did not allow emergency calls, it would cost your company millions of dollars. ___ --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 -- Thx MAG ___ --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] Stopping putgoing calls after working hours
Dear Rich, It seems that my question is very general I apologize for that, but I am glad to see others like yourself pointing me in different directions, it seems all around the world we have problems with the cleaning folks. What I have in mind is to make the phone user lock his phone when he is leaving with a special code and relock it back when he comes to work (and as for emergency calls there are attendants who work at night who will be able to make an emergency call whenever needed at the spot), now there is nothing that seems to be able to do that directly, I have played around with the gotoiftime and also the time based dial plan include sent in mails before that. But while working I thought of another approach why not create a php web interface that each user logs in with a special username and password and gives him access to lock his phone, and what php does is actually change the secret password to something else than the configured on the phone, this should make the phone unable to authenticate thus not being able to make a call, and unlocking it returns the password to it's right form, I have already found the tables that I need to play around so I will restart making the php. I will update the list back with my final result. Do you guys think I could send a mail to the dev site to see if they can add this feature to asterisk. Thx MAG Rich Adamson wrote: > I am trying to find a way to stop people who use phones after business > hours (a policy the company wants to implement), we have cisco 7940 and > 7910 phones and sadly they don't have a phone lock password system (on > these ciscos it locks config menu changes but not the calls but the > cisco 7920 has this feauture). > > So I was wondering is there a way to make this happen in asterisk?? You need to better describe your objectives. If you really mean stop "all" calls (including emergency calls), that's easy. If you mean stop all calls that "cleaning folks" initiate (usually not employees), that just requires some extensions.conf changes to force the user to enter an "access code" before a call can be placed. (Just don't advertise that access code anyone that you don't want making calls. If your talking about a fairly major security issue (such as your users call forwarding their phones to the brother-in-law after normal hours, you'll probably need to disable call forwarding on the phone itself. If your talking about primarily managing expenses, use the CDR detail to generate a personalized report for each employee show this calls make between 5pm and 7am, and forward that report to each employee (and cc: the manager). That's usually enough to significantly cut those calls. If you don't have a policy relative to use of company assets (phones PC's) for personal use, you might put one together and reference that policy in the morning CDR detail report. (I'm sure at lease some of those calls are likely legitimate calls, so cutting all calls is not likely a workable solution. ___ --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 -- Thx MAG ___ --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] Stopping putgoing calls after working hours
Dear All, I am trying to find a way to stop people who use phones after business hours (a policy the company wants to implement), we have cisco 7940 and 7910 phones and sadly they don't have a phone lock password system (on these ciscos it locks config menu changes but not the calls but the cisco 7920 has this feauture). So I was wondering is there a way to make this happen in asterisk?? -- Thx MAG ___ --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] Stopping putgoing calls after working hours
Dear Moj,
Thanks a lot fo the tip, it seems I can do that it is very flexible
and easy to use, I will try to add it to the trixbox files in a nice fashion
but that will be after I get some sleep ;-)
Thx
MAG
"Mojo with Horan Company, LLC" wrote:
Sure, in the context the phones live in, play around
with the
GotoIfTime() application:
Completely pseudocoded, will not work without research:
[internal]
priority 1 : gotoiftime(8:00-17:00|mon-fri?priority 3)
priority 2 : goto 10
priority 3 : dial(out_trunk, ${EXTEN})
priority 4 : hangup
priority 10: play a message "outgoing call restricted"
priority 11: hangup
The next move in your text adventure might be "Show Application
GotoIfTime" from the CLI :)
Moj
Mohamed A. Gombolaty wrote:
> Dear All,
>
> I am trying to find a way to stop people who use phones after
business
> hours (a policy the company wants to implement), we have cisco 7940
and
> 7910 phones and sadly they don't have a phone lock password system
(on
> these ciscos it locks config menu changes but not the calls but the
> cisco 7920 has this feauture).
>
> So I was wondering is there a way to make this happen in asterisk??
>
> --
> Thx
> MAG
>
> !DSPAM:500,4534119649042068143078!
>
>
>
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> !DSPAM:500,4534119649042068143078!
--
Mojo [EMAIL PROTECTED]>
Office Manager, Horan Company, LLC
(907) 747- x112
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MAG
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Re: [asterisk-users] How do you like TrixBox?
Dear All, I am have experimented asterisk long before any gui was available and also currently working with trixbox, ofcourse working with asterisk directly makes you more aware but when you start deploying the system you will face management issues for asterisk, as anyone who deals with asterisk must be experienced enough with it and that will make the people who support the users a few, while with trixbox those few people can be left as escalation points and through GUI you can make other less aware of asterisk administer the day to day tasks. Trixbox in my belief is making more people everyday depend on asterisk ofcourse knowing how to deal directly with asterisk will be a plus but yet this could come by time with trix box and everyday experience being gained will make them someday reach that level. Trixbox is a great start point to implement asterisk but learning asterisk configs must also be in schedule to maintain a persistent environment. Thx MAG Dovid B wrote: Yes but they will never understand the configs. They need to learn step by step. - Original Message - From: "joe, at j4computers" [EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com> Sent: Friday, October 13, 2006 4:11 PM Subject: Re: [asterisk-users] How do you like TrixBox? Dovid B[EMAIL PROTECTED]> Wrote on: 10/13/2006 9:51 AM: >. . . A)If something goes wrong they wont know where to > start. They only know the GUI. B)They will never know the "real way" of > working asterisk.. . . > But, can't it be one way of "learning"? Can't one setup and modify a Trixbox setup, then peruse the conf files, to get familiar with (almost) all things Asterisk? Spoke as one who was not very pleased with their own foray into Trixbox and is still creeping up to speed on Asterisk. joe ___ --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 -- Thx MAG ___ --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] Make Asterisk server initiate a Call
Dear All, We need to do the following crazy scenario which is really stupid but wanted :-((, I need to make the sip server initiate a call on zap channels and once the phone answers, it should play an IVR and according to the choice of the called he will be moved to other extensions, we plan to make an e-mail to trigger that call but I only need to know what commands be used to make the server initiate a call ? I have found some people saying they have done something of this sort but no specific details on configuration, and some talk about a problem in making sure if the call is answered on the other side or not, your help will be very much appreciated. -- Thx MAG ___ --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] Active Directory Listing Feauture
Dear All, I am currently very stumped on the subject of Active Directory listing, as I am unable to find any documents regarding this feature thus I am unable to configure it or know how to use it. Does anyone have any useful info or documents regarding this feature in terms of how to or guides I will be very much thankful. -- Thx MAG ___ --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] Strange Error when calling
Dear All, After doing the test everything went fine, Thanks Anthony for putting me on the right direction. Thx MAG "Mohamed A. Gombolaty" wrote: Dear Anthony, I believe you where right the dial plan seems to have been missing the TRUNK= statement and I found one in the file extensions.conf but not the correct group I configured so I changed it and will test again. Thx MAG "Mohamed A. Gombolaty" wrote: Dear Anthony, The dial plan is currently very simple it should pick up any call and send it to a sip phone registered, you can see the context below named zap-in is what I am using, it is only that and nothing more, is there something extra I have to add to dial plan or to that context ? Thx MAG Anthony Rodgers wrote: This looks like a dialplan problem - do you have a match for 0109687348 in the zap-in context in your dialplan? A. On 26-Jul-06, at 2:40 PM, Mohamed A. Gombolaty wrote: > Dear All, > I have a strange problem in recieving calls on the pri the zaptel > is green and everything seems very well, but when a call comes I > can see the call along with the caller ID but then I get this > strange message which make the call hungup: > > > error msg: 'zap-in' from '0109687348' does not exist. Rejecting > call on channel 0/18, span 1. > > the PRI is an E1 and I have the following configuration for > extensions.conf > > [zap-in] > exten => s,1,Answer > exten => s,2,Dial(sip/100) > exten => s,3,Hungup > > as for the zapata.conf it is as follow: > > [channels] > language=en > switchtype=euroisdn > signalling=pri_cpe > context=zap-in > group=0 > channel=>1-15,17-31 > > I don't know what the problem is or where to look, I will > appreciate it if someone can help me out? > > Thx > MAG > > -- Thx MAG > ___ > --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 -- Thx MAG ___ --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 -- Thx MAG -- Thx MAG ___ --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] Strange Error when calling
Dear Anthony, The dial plan is currently very simple it should pick up any call and send it to a sip phone registered, you can see the context below named zap-in is what I am using, it is only that and nothing more, is there something extra I have to add to dial plan or to that context ? Thx MAG Anthony Rodgers wrote: This looks like a dialplan problem - do you have a match for 0109687348 in the zap-in context in your dialplan? A. On 26-Jul-06, at 2:40 PM, Mohamed A. Gombolaty wrote: > Dear All, > I have a strange problem in recieving calls on the pri the zaptel > is green and everything seems very well, but when a call comes I > can see the call along with the caller ID but then I get this > strange message which make the call hungup: > > > error msg: 'zap-in' from '0109687348' does not exist. Rejecting > call on channel 0/18, span 1. > > the PRI is an E1 and I have the following configuration for > extensions.conf > > [zap-in] > exten => s,1,Answer > exten => s,2,Dial(sip/100) > exten => s,3,Hungup > > as for the zapata.conf it is as follow: > > [channels] > language=en > switchtype=euroisdn > signalling=pri_cpe > context=zap-in > group=0 > channel=>1-15,17-31 > > I don't know what the problem is or where to look, I will > appreciate it if someone can help me out? > > Thx > MAG > > -- Thx MAG > ___ > --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 -- Thx MAG ___ --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] Strange Error when calling
Dear Anthony, I believe you where right the dial plan seems to have been missing the TRUNK= statement and I found one in the file extensions.conf but not the correct group I configured so I changed it and will test again. Thx MAG "Mohamed A. Gombolaty" wrote: Dear Anthony, The dial plan is currently very simple it should pick up any call and send it to a sip phone registered, you can see the context below named zap-in is what I am using, it is only that and nothing more, is there something extra I have to add to dial plan or to that context ? Thx MAG Anthony Rodgers wrote: This looks like a dialplan problem - do you have a match for 0109687348 in the zap-in context in your dialplan? A. On 26-Jul-06, at 2:40 PM, Mohamed A. Gombolaty wrote: > Dear All, > I have a strange problem in recieving calls on the pri the zaptel > is green and everything seems very well, but when a call comes I > can see the call along with the caller ID but then I get this > strange message which make the call hungup: > > > error msg: 'zap-in' from '0109687348' does not exist. Rejecting > call on channel 0/18, span 1. > > the PRI is an E1 and I have the following configuration for > extensions.conf > > [zap-in] > exten => s,1,Answer > exten => s,2,Dial(sip/100) > exten => s,3,Hungup > > as for the zapata.conf it is as follow: > > [channels] > language=en > switchtype=euroisdn > signalling=pri_cpe > context=zap-in > group=0 > channel=>1-15,17-31 > > I don't know what the problem is or where to look, I will > appreciate it if someone can help me out? > > Thx > MAG > > -- Thx MAG > ___ > --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 -- Thx MAG ___ --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 -- Thx MAG ___ --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] bugs.digium.com
Dear All, I just wanted to comment on this point of the discussion: > In a PRODUCTION environment, you can't be running a sip debug to your > console. In a PRODUCTION environment you have all of these issues worked out in your test lab before deploying to production. I do agree with Douglas that having a way to log the debug of sip to a file would be a great option available to use in production, you cannot test a problem occuring to a production system in the lab or even expect problems before going into production to resolve them in the lab, I believe the russel didn't understand well what the request was. But I do hope you can file the bug and really make it obvious that it's a feature request, and I believe someone will take care of it. Thx MAG Andrew Kohlsmith wrote: On Thursday 27 July 2006 10:32, Douglas Garstang wrote: > It clearly is a bug, or at the VERY least, a limitation that needs to be > fixed. So why the hell did he give me -2 karma points and say 'not actually > a bug'. Fine... so how do you file an enhancement request then? If there's > no way to file an enhancement request, then this is the most appropriate > place to file this. When I report a bug, I can say it's for a "Feature Request". Perhaps that's what you should have done? > Its damn irritating not being able to have 'sip debug' output go to a file > only, and this is what the options in logger.conf imply you should be able > to do, which is another reason I don't understand why he took this > irrational action. It's perfectly rational. You posted a bug that is at best a feature request. That's where the -2 came from. I agree with you in the sense that it should not have been closed but simply readdressed, but that's not my call. > In a PRODUCTION environment, you can't be running a sip debug to your > console. In a PRODUCTION environment you have all of these issues worked out in your test lab before deploying to production. -A. ___ --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 -- Thx MAG ___ --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] E1 connectivity question
Dear All, I have bought a digium TE205p in order to move our E1 pri from a siemens pbx to an asterisk server platform, I have already gathered the data needed to configure the card but I am troubled by one thing that seems unclear on all the documents I read. The E1 is currently inserted in a modem and from the modem goes out a cable to the siemens pbx so should I take the E1 from that modem or take the E1 directly from the provider, plus is there any special pin assignment. Your Help will be very much appreciated. -- Thx MAG ___ --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] E1 connectivity question
Dear Steve, Yes I did mean a csu/dsu I will try your suggestion and update the results. Thx MAG Steve Totaro wrote: Mohamed A. Gombolaty wrote: Dear All, I have bought a digium TE205p in order to move our E1 pri from a siemens pbx to an asterisk server platform, I have already gathered the data needed to configure the card but I am troubled by one thing that seems unclear on all the documents I read. The E1 is currently inserted in a modem and from the modem goes out a cable to the siemens pbx so should I take the E1 from that modem or take the E1 directly from the provider, plus is there any special pin assignment. Your Help will be very much appreciated. -- Thx MAG If you really mean to say modem then what you are doing will not work. Maybe you mean a CSU/DSU? If it is a CSU/DSU or the box that the telco owns, take the cable coming out of it. Plug it into your asterisk box and see if you get a green light. I suspect you will since it is working with your Siemens box. If not, make an E1/T1 crossover cable. Pinout is: 1 - 4 2 - 5 Thanks, Steve Totaro Thanks, Steve ___ --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 -- Thx MAG ___ --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] Strange Error when calling
Dear All, I have a strange problem in recieving calls on the pri the zaptel is green and everything seems very well, but when a call comes I can see the call along with the caller ID but then I get this strange message which make the call hungup: error msg: 'zap-in' from '0109687348' does not exist. Rejecting call on channel 0/18, span 1. the PRI is an E1 and I have the following configuration for extensions.conf [zap-in] exten => s,1,Answer exten => s,2,Dial(sip/100) exten => s,3,Hungup as for the zapata.conf it is as follow: [channels] language=en switchtype=euroisdn signalling=pri_cpe context=zap-in group=0 channel=>1-15,17-31 I don't know what the problem is or where to look, I will appreciate it if someone can help me out? Thx MAG -- Thx MAG ___ --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] E1 connectivity question
Dear Steve, The line has worked like charm, but now I am facing a new problem with recieving the call, I have sent another mail with this issue. Thank you very much for your support Thx MAG "Mohamed A. Gombolaty" wrote: Dear Steve, Yes I did mean a csu/dsu I will try your suggestion and update the results. Thx MAG Steve Totaro wrote: > Mohamed A. Gombolaty wrote: > > Dear All, > > > > I have bought a digium TE205p in order to move our E1 pri from a > > siemens pbx to an asterisk server platform, I have already gathered > > the data needed to configure the card but I am troubled by one thing > > that seems unclear on all the documents I read. > > > > The E1 is currently inserted in a modem and from the modem goes out a > > cable to the siemens pbx so should I take the E1 from that modem or > > take the E1 directly from the provider, plus is there any special pin > > assignment. > > > > Your Help will be very much appreciated. > > > > -- > > Thx > > MAG > > > If you really mean to say modem then what you are doing will not work. > Maybe you mean a CSU/DSU? If it is a CSU/DSU or "the box that the telco > owns, take the cable coming out of it. Plug it into your asterisk box > and see if you get a green light. I suspect you will since it is > working with your Siemens box. If not, make an E1/T1 crossover cable. > Pinout is: > 1 -> 4 > 2 -> 5 > > Thanks, > Steve Totaro > > Thanks, > Steve > ___ > --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 -- Thx MAG -- Thx MAG ___ --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] A problem in recieving voice on one side
Dear All, I am having a problem in a scenario I am doing, I have two branches, every branch has has an [EMAIL PROTECTED] that deals with each branch locally and a trunk connected to a central asterisk, now if any branch wants to call another branch it goes from the local asterisk@ home --> to the central asterisk server and then forwarded --> to the remote [EMAIL PROTECTED] server --> to the phone, this works and rings and the call is up but the problem lies in that one side can hear the voice and sends voice but the other side can send voice and not hear anything coming, any ideas where to begin, I would like to highlight some data below: [EMAIL PROTECTED] latest version on both sides. Central asterisk uses Asterisk 1.2.1. phones support reinvite some I am using reinvite=yes If you need any more data I will supply it, I wasn't sure what to put or even where to start, and I didn't want it to be a very long mail. -- Thx MAG ___ --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] Multiple Outbound SIP Trunks
Hi all, In case you have a number of trunks there is a software named astbill (www.astbill.com) in which you can configure the trunks and decide their costs and it will automatically choose the most suitable trunk. Thx MAG Pikoro wrote: By "trunk" I mean each trunk is a different account on the same SIP provider. Yes, they only allow one call per account. We are an internet provider so I can obtain as many trunks(accounts) as I need. Cheers asterisk wrote: On Tue, 15 Nov 2005, Pikoro wrote: There will be no discrimination or routes based on outbound calling, like a certain trunk for international calls, another for local calls, etc... Only a group of 10 SIP trunks to be rotated for all outbound calls. Can you explain what you mean by a "SIP trunk"? I took it to mean different accounts or providers. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error compiling asterisk addons version 1.2.0-beta2
Dear All, I am facing a problem in compiling the add-ons for the mysql, though the files are downloaded correctly and checked and I tried different mirrors even the cvs but yet I get those errors : app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory cdr_addon_mysql.c:38:19: mysql.h: No such file or directory cdr_addon_mysql.c:39:20: errmsg.h: No such file or directory res_config_mysql.c:51:19: mysql.h : No such file or directory res_config_mysql.c:52:27: mysql_version.h: No such file or directory res_config_mysql.c:53:20: errmsg.h: No such file or directory anyone has a clue, I used to compile it without problems Thx MAG ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Admission Control in Asterisk
Dear All, I was trying to limit the number of calls between different located sites in order to avoid congestion of the bandwidth, but as I found from the mails and testing that it is easy to do it for the incoming calls by the setgroup() and group_count while it is the outgoing is hard to track or limit, So I was wondering if we will see a Call Admission Control soon in Asterisk that can do this job or not? Thx MAG ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Admission Control in Asterisk
Hi Trixter, Yes i did try to make setgroup for the outbound but the problem is after you move it to the desired context or extension in the gotoif statement the group that you have set it in is back to zero so I really can't use it for the outbound, the group used for the outbound will not give the correct count of users dialling out. As u said I am using CVS-Head and used the group_count() with gotoif statements so I am clear of the checkgroup() bug. Thx MAG trixter aka Bret McDanel wrote: On Sun, 2005-10-23 at 11:42 +0200, Mohamed A. Gombolaty wrote: > Dear All, > > I was trying to limit the number of calls between different located sites in > order to avoid congestion of the bandwidth, but as I found from the mails and > testing that it is easy to do it for the incoming calls by the setgroup() and > group_count while it is the outgoing is hard to track or limit, So I was > wondering if we will see a Call Admission Control soon in Asterisk that can do > this job or not? setgroup() should work for outbound. Did you try it and have problems? In asterisk 1.0.x there is a bug about transfered calls, is that where you were running into problems? I find this unlikely since you referenced group_count, which is a 1.2 function (replacing the deprecated checkgroup()). http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 Name: signature.asc signature.asc Type: application/pgp-signature Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Goto command question
Dear Eric,
You are totally right, I already know the information below
but I don't know why I couldn't see them, I certainly need a vacation,
anyway it worked like charm.
Thx
MAG
Eric \"ManxPower\" Wieling wrote:
Mohamed A. Gombolaty wrote:
> Dear All,
>
> I have this question regarding goto command, I amusing Asterisk cvs
head
> version, and I am trying to put a goto statement to send the user
to
> another extension that contains the extension he is dialing
here is how I
> am doing it :
>
> exten => 2x.,1,setgroup(outgoing)
> exten => 2x,2,checkgroup(2)
> exten => 2x.,3,goto(another-context, ${EXTEN},1)
> exten => 2x.,104,hangup
>
> but the result is always it hangs up I don't know if this goto statement
is
> correct or not, can anyone lead me to the right way to make this
statement?
First of all patterns must start with _
exten => _2X.,1,setgroup(outgoing)
Second you are using different patterns
exten => _2X.,1,setgroup(outgoing)
Is NOT the same as
exten => _2X,2,checkgroup(2)
The first pattern is _2X. the second pattern is _2X
Third, do not put spaces after commas.
Try this:
exten => _2X.,1,SetGroup(outgoing)
exten => _2X.,2,CheckGroup(2)
exten => _2X.,3,Goto(another-context,${EXTEN},1)
exten => _2X.,104,Hangup
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--
Thx
MAG
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[Asterisk-Users] Goto command question
Dear All,
I have this question regarding goto command, I amusing Asterisk cvs
head version, and I am trying to put a goto statement to send the user
to another extension that contains the extension he is dialing here
is how I am doing it :
exten => 2x.,1,setgroup(outgoing)
exten => 2x,2,checkgroup(2)
exten => 2x.,3,goto(another-context, ${EXTEN},1)
exten => 2x.,104,hangup
but the result is always it hangs up I don't know if this goto statement
is correct or not, can anyone lead me to the right way to make this statement?
--
Thx
MAG
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Re: [Asterisk-Users] cdr_mysql does not write to mysql db
Dear Kib, As I believe the Realtime options concerning the mysql database can only be used with the Asterisk CVS-HEADversion it's still not implemented on Asterisk v 1.0.* . Thx MAG Kib Eki wrote: Hi, I configured cdr_mysql (addons 1.0.9) to write the cdr records to the mysql db. The problem is that no records are written to the db. Why? I can import the csv-file to the db. so i assume the db is setup correct. Is there any chance to get debug from cdr_mysql to find his problem? This is my cdr_mysql.conf file: [global] hostname=localhost dbname=cdr password=passw0rd user=root ;port=3306 ;sock=/tmp/mysql.sock userfield=1 Thanks and Regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems installing asterisk-addons
Hi Angus, I don't believe it can be the root password of mysql, I used to install the addons without even haved installed mysql server yet, I guess we need to know which platform are you working on and which version you are trying to install. Thx MAG Angus Comber wrote: Hello I have downloaded asterisk-addons but when I make install get: cc -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 I have set a password for root on mysql - could that be the problem? Should I remove the password? What is easiest way to do that? Angus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with Realtime registration problem
Dear All, I am currently working on asterisk cvs-head version in order to use realtime with mysql, 2 asterisk servers with duplicate mysql databases, one asterisk server is serving the sip phones and the data is logged to the database and replicated to the other asterisk database, when the first server fails though it has the sip phones data in it's database the sip phones need to re-register again to work, I am confused as I thought the realtime option should solve this problem since it can use the data stored in it's database. I also tried with the sip.conf the following options: rtcachefriends=yes rtnoupdate=yes rtautoclear=yes rtignoreexpire=yes but with no success, I also tried a suggestion to do a show database command on asterisk cli but that didn't change anything, I was wondering if anyone has notes or ideas regarding this issue I will be very thankful. Thx MAG -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Vovida Loadbalancer
Dear All, I was trying to load balance between two asterisk servers using vovida.org loadbalancer, but when I was running it i faced the following problems: -When phones try to register the lpproxy gives the following message for reach phone trying to connect: Sticky header data is: Call-ID: [EMAIL PROTECTED] No proxies are up - can not send message to anyone (I start the process by ./lbproxy -proxy asterisk1 -proxy asterisk2 ) and the phne is unable to register - doing a sip debug on both asterisk boxes I can see this message on both of them: find_call: Call missing call ID from 'loadbalancerserver' Did anyone face this before or worked with both asterisk and vovida loadbalancer ? Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk realtime failover problems
Dear All, I was trying to use Realtime Asterisk option for Sip users and peers + Heartbeat + Mysql Replication in order to make a failover system, so that if Ast1 went down for any reason, Ast2 server will have the same data that Ast1 used in the Mysql database and don't need to make the phones re-register. But when I started testing: the calls that where active during the transition between the two servers where disconnected (the two phones are talking peer to peer thanks to the canreinvite option but they we still sending UDP packets to port 1025 to the asterisk server), the phones must re-register with the new server though the Mysql server was replicated and the new server should have the data it needs. Has anyone trid doing this before, or does anyone have any idea if this should work or is there another way to do so, I will really appreciate it very much if anyone has any helping pointers. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk realtime failover problems
Dear All, I was trying to use Realtime Asterisk option for Sip users and peers + Heartbeat + Mysql Replication in order to make a failover system, so that if Ast1 went down for any reason, Ast2 server will have the same data that Ast1 used in the Mysql database and don't need to make the phones re-register. But when I started testing: the calls that where active during the transition between the two servers where disconnected (the two phones are talking peer to peer thanks to the canreinvite option but they we still sending UDP packets to port 1025 to the asterisk server), the phones must re-register with the new server though the Mysql server was replicated and the new server should have the data it needs. Has anyone trid doing this before, or does anyone have any idea if this should work or is there another way to do so, I will really appreciate it very much if anyone has any helping pointers. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asking again
Hi Wasim, Check out the x-lite softphone http://www.xten.com/ As for linux check this page there are two softphones type available : http://www.iptel.org/products Thx MAG wassim Darwish wrote: ok what softphone i should use to fit windows and linux supporting iax,thanks in advance. _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Realtime database Problem
Dear Matt, Yes indeed I did I have used cvs to download asterisk and it's addon from CVS. Thx MAG Matthew Boehm wrote: Did you install res_config_mysql.so from asterisk-addons? -Matthew > From: "Mohamed A. Gombolaty" [EMAIL PROTECTED]> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > asterisk-users@lists.digium.com> > Date: Sun, 10 Jul 2005 12:16:51 +0300 > To: Asterisk Users Mailing List - Non-Commercial Discussion > asterisk-users@lists.digium.com> > Subject: [Asterisk-Users] Asterisk Realtime database Problem > > Hi All, > > I am facing a problem with makeing asterisk work realtime with mysql, after > following the tiki steps which are: > > uncommented the lines sipuser and sippeers from extconfig.conf > copied the res_mysql.conf and configured it with the right parameters > checked that mysql is working > added the realtime switch to the extensions.conf > > Now when asterisk is starting I don't see it even to attempt to parse the > res_mysql.conf file so I am assuming that there is something missing what is > it I > don't know. > > -- > Thx > MAG > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Realtime database Problem
Hi All, I am facing a problem with makeing asterisk work realtime with mysql, after following the tiki steps which are: uncommented the lines sipuser and sippeers from extconfig.conf copied the res_mysql.conf and configured it with the right parameters checked that mysql is working added the realtime switch to the extensions.conf Now when asterisk is starting I don't see it even to attempt to parse the res_mysql.conf file so I am assuming that there is something missing what is it I don't know. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Fwd: Asterisk Balancing solution]
Dear All, I read your notes and was very glad, it was a healthy and useful debate, I have set my mind on implementing Realtime for sipusers and peers with mysql database and either use the Mysql replication process or mount the database on both servers. I will write a document of this trial and post it when I finish it out, but during research I found this link and before anything it says that realtime will not work with Asterisk 1.0.7 is that true cause that's what I am using, the second thing was in the extconfig file, can i do something like this sip.conf => mysql,mysqlserver:asteriskdatabase,table Sorry for being late but I was off the last two days due to sickness. Thx MAG Michiel van Baak wrote: On 15:21, Thu 30 Jun 05, Erik Espinoza wrote: > I can only think of 2 ways to proceed: > > 1) Set a shorter register interval > 2) Set static ip on all phones, and forgo registration > 2 should work. We have dynamic addresses on our local net, but the dhcp server gives an address for 30 days to the phones. I did setup the host=dynamic and defaultip=phone.ip.address.with.first.reg This allows me to place calls to no longer registered phones (specially the GS BT 100 looses registration lots of times) Maybe this is of any help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Failover question
Dear All, I am using Linux-High Availability between two Asterisk servers, everything is fine but I do have one problem with this, When a server fails and the other assumes the ip address and start asterisk on server 2, the ip phone must re-register themselves again, otherwise the phones won't ring. Does anyone have Ideas of how to overcome this. Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk failover solution
Dear All, I am using Linux-High Availability between two Asterisk servers, everything is fine but I do have one problem with this, When a server fails and the other assumes the ip address and start asterisk on server 2, the ip phone must re-register themselves again, otherwise the phones are dead. Does anyone have Ideas of how to overcome this. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Fwd: Asterisk Balancing solution]
Dear All, I am using Linux-High Availability between two Asterisk servers, everything is fine but I do have one problem with this, When a server fails and the other assumes the ip address and start asterisk on server 2, the ip phone must re-register themselves again, otherwise the phones are dead. Does anyone have Ideas of how to overcome this. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with failover and load balancing
Dear All, I was searching voip-info for Failover and load balancing for Asterisk, my goal here is to have a system where the SIP traffic is being divided on five central servers with Asterisk on, and if an asterisk server fails another asterisk server will assume it's place , from my readings I have cited the following options: 1- SER + ASTERISK with Domain SRV 2- vovida Load balancer (I am not happy about this one it's old I can't compile on new OSand it's mailing list is useless and development seems to have stopped ) I hope any one could enlighten me with his experience if he has done such a thing and which can be a better option or if there is something I am still missing. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call divert to TRUNK ,if one number is unregistered?
Hi Erdem, Can you try to put another dial command that points to the trunk afetr the dial command to the SIP? fro example: exten => XXX,1, dial(sip/,20,r) exten => XXX,2,dial(zap/) -> note here that I am not sure if the order number should be 2 or 102 but if this didn't work try the other one. Thx MAG Erdem HAKÝ wrote: I have a question. I have two numbers on Asterisk like 902121234567 and 902123645789 and i want to divert first number's call to Trunk if second number is unregistered. Is it possible? Ýf yes, how? Flow Diagram: *Two numbers are registered on Asterisk 902121234567 registered to Asterisk 902123645789 registered to Asterisk *One number is registered, other one is not registered 902121234567 registered to Asterisk 902123645789-x not registered to Asterisk *So first number want to make a call second one (desired situation) 902121234567> Asteriskà Trunk Thanks for your interest. Erdem HAKI - [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER and Asterisk question
Dear All, I just solved my problem, you can make asterisk itself make phones peer with each other if they support the canreinvite option, so step number one was to insert this option in the sip.conf configration in the phones part: [xxx] canreinvite=yes Now the calls will be direct between the two IP Phones without having asterisk in the middle which will save bandwidth on the wan link. As for SER when you perform it after this step it shoild work fine with you. Thx MAG Mohamed A. Gombolaty wrote: Dear Yair, Actually what happens is that from SER debug I can see the call is looping between Asterisk and SER. but adding a number makes no loops. Thx MAG Yair Hakak wrote: yes, there is. run everything through asterisk, no matter how long the extensions are. for example, 666 calls 999 goes to asterisk, sees a dial sip:[EMAIL PROTECTED], goes back to SER. bounces back to ser. If everything is working well asterisk will set up the call and get out of the way. I don't see why you need to prepend digits in order to make this work, if i'm missing something let me know. -yair On 6/16/05, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote: Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and which has a uri rule that says forward 4 digit starting with 1 to the asterisk sip port the asterisk extensions.conf has an entry for 1999 and dials [EMAIL PROTECTED], if not answered voicemail runs and so on. ain't there a way to make 666 directly call 999 without using 1999. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER with Asterisk Problem
Dear All,
I am trying to make my sip phones register with SER and make use of
Asterisk capabilities such as voicemail and parking calls for example.
on SER side
the ip of the server is 192.168.99.170 and uses port 5060
in my ser.cfg I added the following lines :
if (uri=~"sip:[EMAIL PROTECTED]") {
rewritehostport("10.3.26.2:5090");
t_relay();
break;
}
all my sip phones can register to ser without passwords.
On the Asterisk side:
the ip is 10.3.26.2 and uses port 5090
in my sip.conf I added:
register => 10:[EMAIL PROTECTED]/10
[sip-ser}
type=friend
user=10
userfrom=10
host=192.168.99.170
Now My problem :
1- the asterisk console shows failed messages to register to the ser
(Forbidden - wrong password authentication)
--
Thx
MAG
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[Asterisk-Users] SER and Asterisk question
Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and which has a uri rule that says forward 4 digit starting with 1 to the asterisk sip port the asterisk extensions.conf has an entry for 1999 and dials [EMAIL PROTECTED], if not answered voicemail runs and so on. ain't there a way to make 666 directly call 999 without using 1999. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER and Asterisk question
Dear Yair, Actually what happens is that from SER debug I can see the call is looping between Asterisk and SER. but adding a number makes no loops. Thx MAG Yair Hakak wrote: yes, there is. run everything through asterisk, no matter how long the extensions are. for example, 666 calls 999 goes to asterisk, sees a dial sip:[EMAIL PROTECTED], goes back to SER. bounces back to ser. If everything is working well asterisk will set up the call and get out of the way. I don't see why you need to prepend digits in order to make this work, if i'm missing something let me know. -yair On 6/16/05, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote: Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and which has a uri rule that says forward 4 digit starting with 1 to the asterisk sip port the asterisk extensions.conf has an entry for 1999 and dials [EMAIL PROTECTED], if not answered voicemail runs and so on. ain't there a way to make 666 directly call 999 without using 1999. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Xlite not communicating with Asterisk
Dear All, I have downloaded the xlite version 2.0 for windows and I made the following conf in the xlite itself as the document suggested in order to make it work with Asterisk but still it doesn't work as a matter of fact when I tried to make a tcp dump I can see no packets going between the windows client and the Asterisk server at all, here is the my conf on the xlite itself: in the MenuSystem SettingsSIP ProxyDeafult Enabled: yes Display Name: Username: Authorization User: Password: Domain/Realm: mysip.server.com SIP Proxy: 192.168.99.243 Outbound Proxy: Use Outblound Proxy: Default Send internal IP: Always Register: Always Direct Dial IP: NO DIal Prefix: my sip.conf for the device is as follow: [881] ;Turn off silence suppression in X-Lite (Transmit Silence=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend secret= callerid=Mohamed Mahmoud 881 host=dynamic dtmfmode=inband context=from-sip canreinvite=no disallow=all allow=gsm ofcourse I added in the context mentioned above the macro I use with all my extensions. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xlite not communicating with Asterisk
Hi Shahan,
yes both are in the same LAN
Thx
MAG
Shahan Kalutanthri wrote:
HI..!!
Is you windows PC the Asterisk in the same LAN.
-Original Message-
From: Mohamed A. Gombolaty [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, June 08, 2005 2:29 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Xlite not communicating with Asterisk
Dear All,
I have downloaded the xlite version 2.0 for windows and I made the following
conf in the xlite itself as the document suggested in order to make
it work
with Asterisk but still it doesn't work as a matter of fact when I
tried to
make a tcp dump I can see no packets going between the windows client
and
the Asterisk server at all, here is the my conf on the xlite itself:
in the Menu>System Settings>SIP Proxy>Deafult
Enabled: yes
Display Name:
Username:
Authorization User:
Password:
Domain/Realm: mysip.server.com
SIP Proxy: 192.168.99.243
Outbound Proxy:
Use Outblound Proxy: Default
Send internal IP: Always
Register: Always
Direct Dial IP: NO
DIal Prefix:
my sip.conf for the device is as follow:
[881]
;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! ;Note
that
Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend
secret= callerid="Mohamed Mahmoud" 881> host=dynamic dtmfmode=inband
context=from-sip canreinvite=no disallow=all allow=gsm
ofcourse I added in the context mentioned above the macro I use with
all my
extensions.
--
Thx
MAG
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--
Thx
MAG
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Re: [Asterisk-Users] Xlite not communicating with Asterisk
Hi Wilson, yes I am leaving it blank although I did try to use a username in the sip.conf but with the same result also I have tried to put the extension 881 but the same result. Wilson Pickett wrote: > Enabled: yes > Display Name: > Username: > Authorization User: > Password: > Domain/Realm: mysip.server.com Is this your username: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco Softphone 1.3(4a) issue.
Dear All, I was trying to configure Asterisk to work with Cisco Softphone version 1.3(4a) and I am having a problem, the Softphone when is started asks for a Line to use, all documents I found specify this is something to be done from t Cisco Call Manager, has any one worked on this before? -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie :Call Forwarding problem
Dear All,
I was trying to enable call forwarding, following the steps of the link
on voip.org regarding this issue it doesn't work and the phone I am trying
to implement on is still ringing. below is my conf in extensions.conf and
the CLI output during the process.
My configuration is :
exten => _*5X.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:2})
exten => _*5X.,2,Hangup
exten => *5,1,DBdel(CF/${CALLERIDNUM})
exten => *5,2,Hangup
[macro-stdexten]
;
; Standard extension macro (with call forwarding):
; ${ARG1} - Extension(we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,DBget(temp=CF/${ARG1})
exten => s,2,Goto(${temp}|1)
exten => s,102,Goto(s|3)
exten => s,3,Dial(${ARG2},120)
exten => s,103,Goto(s|50)
exten => s,4,Voicemail(u${ARG1})
exten => s,5,Hangup
exten => s,104,Voicemail(b${ARG1}) ; busy
exten => s,105,Hangup
the output on the CLI during this process was:
*CLI>
-- Executing DBdel("SIP/777-a77c", "CF/777") in
new stack
-- DBdel: family=CF, key=777
Urgent handler
-- Executing Hangup("SIP/777-a77c", "") in new stack
Urgent handler
-- Executing DBput("SIP/777-ad46", "CF/777=888")
in new stack
-- DBput: family=CF, key=777, value=888
Urgent handler
-- Executing Hangup("SIP/777-ad46", "") in new stack
Urgent handler
*CLI>
*CLI>
-- Executing Dial("SIP/999-8f50", "SIP/777|7|tr")
in new stack
-- Called 777
Urgent handler
Urgent handler
-- SIP/777-82e9 is ringing
Urgent handler
Any Idea what's wrong
--
Thx
MAG
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Re: [Asterisk-Users] Newbie :Call Forwarding problem
Dear Peter, here is my 777 conf in extensions.conf: [Internal-sip] exten = 777,1,Dial(SIP/777,7,tr) exten = 777,2,Dial(SIP/777SIP/888,10,tr) exten = 777,3,voicemail,u777 exten = 777,104,voicemail,b777 As for the stdexten macro I really don't know what you mean by using it do you mean by doing include = , if this is the case I didn't but if you mean something please tell more. Thx MAG Peter Bowyer wrote: How is extension 777 defined in extensions.conf? Did you use the stdexten macro? Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie :Call Forwarding problem
Hi Peter, You are totally right it worked, and I really loved the macro idea I have mostly grasped it now and will use it more extensivley in the future. Thx MAG Peter Bowyer wrote: On 02/06/05, Mohamed A. Gombolaty [EMAIL PROTECTED]> wrote: > here is my 777 conf in extensions.conf: > [Internal-sip] > > exten => 777,1,Dial(SIP/777,7,tr) > exten => 777,2,Dial(SIP/777SIP/888,10,tr) > exten => 777,3,voicemail,u777 > exten => 777,104,voicemail,b777 > > As for the stdexten macro I really don't know what you mean by using it do you mean > by doing include => , if this is the case I didn't but if you mean something please > tell more. There's your problem, then - you've got a macro to do the call forwarding but you're not using it in the dialplan. Instead of all that for 777, try this: exten => 777,1,macro(stdexten,777,SIP/777) Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Meeting VS Call Confrence
Dear All, I was trying to make call confrence available but all the asterisk documents use the meeting room concept, where those who wanna meet have to dial an extension corresponding to the meeting room, while call conference actually means that I am on exten 100 I can dial exten 200 and add it to confrence and again dial 333 and add it to the confrence and so on. Is there any way to make call confrencing available and not meeting room concepts? -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR Load
Dear All,
I was trying to enable call forwarding, following the steps of the link
on voip.org regarding this issue it doesn't work and the phone I am trying
to implement on is still ringing. below is my conf in extensions.conf and
the CLI output during the process.
My configuration is :
exten => _*5X.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:2})
exten => _*5X.,2,Hangup
exten => *5,1,DBdel(CF/${CALLERIDNUM})
exten => *5,2,Hangup
[macro-stdexten]
;
; Standard extension macro (with call forwarding):
; ${ARG1} - Extension(we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,DBget(temp=CF/${ARG1})
exten => s,2,Goto(${temp}|1)
exten => s,102,Goto(s|3)
exten => s,3,Dial(${ARG2},120)
exten => s,103,Goto(s|50)
exten => s,4,Voicemail(u${ARG1})
exten => s,5,Hangup
exten => s,104,Voicemail(b${ARG1}) ; busy
exten => s,105,Hangup
the output on the CLI during this process was:
*CLI>
-- Executing DBdel("SIP/777-a77c", "CF/777") in
new stack
-- DBdel: family=CF, key=777
Urgent handler
-- Executing Hangup("SIP/777-a77c", "") in new stack
Urgent handler
-- Executing DBput("SIP/777-ad46", "CF/777=888")
in new stack
-- DBput: family=CF, key=777, value=888
Urgent handler
-- Executing Hangup("SIP/777-ad46", "") in new stack
Urgent handler
*CLI>
*CLI>
-- Executing Dial("SIP/999-8f50", "SIP/777|7|tr")
in new stack
-- Called 777
Urgent handler
Urgent handler
-- SIP/777-82e9 is ringing
Urgent handler
Any Idea what's wrong
--
Thx
MAG
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[Asterisk-Users] Ztdummy usage
Dear All, I have installed Asterisk everything is OK until I tried to configure meeting room, configuration was simple enough when I try I get a message that it's not a valid meeting room, Now I don't have a Zaptel device on my machine, so I found that you will have to use ztdummy to make a dummy zaptel device on your machine and this is because of timing issues. My question is ztdummy can only be done when making asterisk or is ther a way to do it after post installation? I am using by the way freebsd 5.3, built Asterisk from the ports successfully. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk compailation Error Chan_zap.c
Dear Ghassan, I never used fedora but in the link below you will find a step by step installation for fedora platform check it out and see if you are missing anything. http://www.voip-info.org/wiki-Asterisk+Linux+Fedora Thx MAG Ghassan Lama wrote: Hi; It is my first time installing an asterisk PBX system ... I do have a TDM400 wildcard with 4 FXO moduls on a PC with 3.0GHZ HT CPU and INTEL 915 moatherboard ... Fedora C2 Linux as O.S. and I have the latest CVS astreisk , Zaptel and Libpri downloaded the zaptel drivers installation and configuration seems to be fine and the libpri but when I tried to compile and install the asterisk software the following error occurred : Chan_zap.c 2772 : error : "Zt_event_DTMFDIGIT" undeclared Can any body help why this error .. Thanks; Ghassan M. Lama' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with another Asterisk
Hi Chris, Did you try the echo test, this will help us to better test the latency between the two distance phones, the link below should guide you through the echo cmd. http://www.voip-info.org/tiki-index.php?page=Asterisk cmd Echo Thx MAG Giles Coochey wrote: > > Has anyone seen a situation where, upon connecting two > asterisk servers > together with IAX registration, outgoing/incoming calls that > route through > both servers are choppy and jittery? I don't have this > problem when I call > out to teliax (my ITSP) directly, but if I try to make the > call through the I found this problem minimised when I used the same codec end-to-end. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP V2 Support
Dear All, I am totally new in this arena and I am still waiting for my installation process on freebsd to finish, but I wanted to make sure of the following: - Call routing between IP telephones can be done regardless of who made the phones? - Asterisk does support SIP V2? - it does act as SIP Proxy and Register? -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
