[Asterisk-Users] choppy music on hold - only on PRI PSTN
Hello to all I do not know what is causing choppy music on hold when call comes in through E1 card (PRI).. but this channel info is somehow strange.. We use Alaw over PRI (and I think its format number 8), But why is WriteFormat at 2 ? Thanks! show channel Zap/1-1 -- General -- Name: Zap/1-1 Type: Zap UniqueID: 1136667936.0 Caller ID: 04573573 Caller ID Name: (N/A) DNID Digits: 349 State: Up (6) Rings: 1 NativeFormat: 72 WriteFormat: 2 ReadFormat: 8 1st File Descriptor: 14 Frames in: 3516 Frames out: 3352 Time to Hangup: 0 Elapsed Time: 0h1m10s Direct Bridge: none Indirect Bridge: none -- PBX -- Context: OZ0800 Extension: s Priority: 7 Call Group: 0 Pickup Group: 0 Application: Queue Data: OZ0800|Tt|||300 Blocking in: ast_waitfor_nandfds ___ --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] ipVolution
Hi, Anybody have some experience and did some testing with ipVolution E1/T1 cards? goran ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to configure two Asterisk servers for onecall center
On Fri, 2005-10-21 at 09:39 -0700, Tielin Xu wrote: Hi All: I have a situation to be resolved. Assume that one location call center with 150 agents. I have two asterisk servers to serve those 150 sip phones. The servers are connected to PSTN as 4 T1/PRI for each. My question is why do you have about 150% the agents to the line capacity? Even with pauses and all do you expect that the 96 (or less in the case of pri) lines to be in use at all times? Predictive dialing ? ___ --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] Voicemail as an email attachement
Small.. just app_voicemail.c and a sendEmail script... You can download it from here: app_voicemail.c http://www.migo-systems.com/index.php?option=com_remositoryItemid=11func=f ileinfoid=9 and sendEmail http://www.migo-systems.com/index.php?option=com_remositoryItemid=11func=f ileinfoid=10 sendEmail is most important.. code change is really small in app_voicemail.. but here it is.. 1. install sendEmail 2. Edit app_voicemail.c : You will need to change app_voicemail.c to suit your needs.. Go to line 1035 (or find goran.skular) and: Change [EMAIL PROTECTED] to from address you want to show up Mail.slsolucije.hr:25 change to your mail.server.xxx:smtp Password_here is place for your password.. Go to line 1130 also (or find next appereance of goran.skular) and to the same again. That's all in short. Have a nice day. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, October 19, 2005 4:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail as an email attachement Yes. I am interested. I will make provisions for the upload. How big are the files? Thanks BEN Goran Skular wrote: I changed my app_voicemail.c to work not with sendmail but with sendEmail that connects to any SMTP and sends email with attachment... It's dirty, but it works. If you are interested I can upload app_voicemail.c and sendEmail package somewhere.. I have configured the voicemail.conf file as per the wiki to email voicemails as an attachment. I cannot find any instructions/locations to set the outgoing server login information. Furthermore, I can get no emails from asterisk. Can anyone point me to the next step to setup the attachment of voicemail messages to an email? Thanks BEN ___ --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 ___ --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
RE: [Asterisk-Users] Voicemail as an email attachement
On Tue, Oct 18, 2005 at 11:22:52PM -0500, Ben Brown wrote: I have configured the voicemail.conf file as per the wiki to email voicemails as an attachment. I cannot find any instructions/locations to set the outgoing server login information. Furthermore, I can get no emails from asterisk. Can anyone point me to the next step to setup the attachment of voicemail messages to an email? Set up a sendmail. Or basically: an MTA. Any linux distro comes with at least one (postfix seems to be the preffered choice nowadays). Which one do you use? There are a bunch of programs that provide /usr/sbin/sendmail but don't spool the result. Check msmtp, ssmtp, masqmail and nullmailer. There are probably others. The downside is that messages that have, for some reason, not been delivered in the first shot (e.g: due to some transient network error) will be dropped rather than queued. I was playing with mta, but this is so complicated, specially if you are on dynamic ip address, so it is much easier to use smtp for sending mails.. ___ --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] Asterisk in Croatia - Zagreb
Hello, I was wondering how many people from Croatia are using and playing with Asterisk. Recently I had a contact with one user and I am very glad. It will be really nice to organize a Croatian Asterisk community and on that way we are organizing a little gathering. It does not matters how much experience you have, everthing you need is some interest in Asterisk. Beside my last contact I know that croatian wifi community ZG Wireless is using Asterisk also. So, Everyone of you, located in Croatia, please contact me here or on email. For the purpose of collecting as much people, gathering is to be expected next month (around 19th) Send me an e-mail or even register on www.migo-systems.com. Further info will be available later. Looking forward for it, Goran Skular www.slsolucije.hr ___ --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] Voicemail as an email attachement
On Thu, Oct 20, 2005 at 08:58:01AM +0200, Goran Skular wrote: I was playing with mta, but this is so complicated, specially if you are on dynamic ip address, so it is much easier to use smtp for sending mails.. Sending is never a problem. Recieving is a problem when you're on a dynamic address. You can tell your MTA to do just that. e.g, on postfix, in /etc/postfix/main.cf: # assuming a well-behaved setup relayhost = the.isp.domain # and if not: relayhost = [smtp.the.isp.domain] BTW: one option you have with a decent mailer is not to write the email address in voicemail.conf, but rather, to write there for each box the email vmbox-vmbox, and use the MTA's aliases to map them to emails. Either using a plain text /etc/aliases, or using any other database (ldap, mysql, whatever). If relaying is enabled and accepted on remote side... and nowdays is hard to enable relaying with those spammers around.. I tried something with this relaying, but without success, so I changed app_voicemail in order to send mail with SMTP and sendEmail script. Can you tell me how to accept relaying on server, but to limit it to allowable IP address (which is in this case dynamic ip..). That will help me a lot :) ___ --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] Asterisk in Croatia - Zagreb
Hello, I'm there with you, dude, haven't talked to you in some 5-6 years? :) I know a couple of people that are working with Asterisk... Cheers, Vedran. Nice surprise ! :) Ok, you're the first participant along with me on this small gathering. I sent you email, and let's ring on those guys you know. I hope that we will find some people out there for a nice gathering on that subject (and subjects involved in our past 5-6 years you mentioned :) ) See you, Goran ___ --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] New ISDN architecture available for asterisk
Hi to all, sorry for crossposting the -dev and -user lists, but I think this could be quite interesting news for EuroISDN people, expecially BRI owners. A new ISDN architecture, called vISDN, has been developed to fully support EuroISDN protocol with HFC based cards: HFC-S PCI, HFC-4S and HFC-8S (with HFC-E1 and HFC-S USB support coming soon). vISDN is not based on Zaptel, libpri, chan_zap, zaphfc, qozap, etc... but has been designed from scratch to be a standard compliant EuroISDN implementation plus a channel crossconnector, plus protocol analisys support thru Ethereal, plus a ppp terminator, plus other stuff :) Very, very nice.. I am looking forward for test it. Further, I hope that ecgo cancelation will be implemented also in near future, as it is very important in most cases. Are there maybe some HFC (both BRI and PRI) boards with hw echo cans, or they are all passive? For small Euro BRI installations we are using at this moment HFC with bristuff. But where E1 is involved, we are trying now to avoid E1 cards without HW echo cans integrated. At this point we are considering between Sangoma and Digium with hw cans... but who knows what HFC boards would bring. Beronet and Junghanns are here to be observed.. Kind regards, Goran ___ --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] Voicemail as an email attachement
Always download programs directly from the homepage or from another reliable source. Don't just grab programs and scripts from everywhere. But why not just set mailcmd in voicemail.conf? Also, quoting the homepage: Why not use sendmail? Sendmail is a large and complex mail server. Installing this kind of mail software on servers (unless it's a mail server) is more of a security risk than its worth. Not if it only listens on localhost or doesn't listen at all. The codebases of sendmail is indeed known to be a source of many security breaches, but exim, postfix and qmail are not so. Most distros come with either postfix or exim by default nowadays. Not to mention it can be a real pain messing with configuration files and such. Systems need another simpler way to send email from the command prompt, and sendEmail provides this functionality. Its a simple, direct way to send email without the overhead of other conventional email software. Most of the pain is caused due to management of messages in the queue. Other types of pain are due to messages routing. Routing issues can be easily solved by sending basically all mail to a remote host (excpt, maybe, some system messages). However, if the system is disconnected from the net for a while what will you do? lose all voicemail messages? (and get just ugly warnings in the logs as a reminder) Also note that there are quite a few programs that could use a sendmail-compatible interface. cron sends its output using mail. So are many other programs. If you don't provide a sendmail-compatible interface (even if it one that does not queue, something like nullmailer) you'll have to reconfigure other parts of your system as well. And worst of all: you won't be able to send mail with mutt. The horors! I completely agree! This is only a work-around.. there are much better methods involved with sendmail which is really powerfull and thus really complicated to configure. The most difficult part is not on * server side, but on relaying server side which must be configured to allow relays only from authorized sites.. I had no success with that, even with some keys and similar solutions which I tried, so I gave up and start using sendEmail. But, I will for sure migrate to sendmail when time for that comes, and I strongly suggest it to everyone. ___ --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] Asterisk in Croatia - Zagreb
At this moment we are counting 4 possible participants. (Appoligies for those who are not from Croatia for using this list, but this list has a lot of subscribers including from Croatia) We are waiting for others to join us. Feel free to respond here or on my e-mail. Thanks! Hello, I was wondering how many people from Croatia are using and playing with Asterisk. Recently I had a contact with one user and I am very glad. It will be really nice to organize a Croatian Asterisk community and on that way we are organizing a little gathering. It does not matters how much experience you have, everthing you need is some interest in Asterisk. Beside my last contact I know that croatian wifi community ZG Wireless is using Asterisk also. So, Everyone of you, located in Croatia, please contact me here or on email. For the purpose of collecting as much people, gathering is to be expected next month (around 19th) Send me an e-mail or even register on www.migo-systems.com. Further info will be available later. Looking forward for it, Goran Skular www.slsolucije.hr ___ --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] Voicemail as an email attachement
I changed my app_voicemail.c to work not with sendmail but with sendEmail that connects to any SMTP and sends email with attachment... It's dirty, but it works. If you are interested I can upload app_voicemail.c and sendEmail package somewhere.. I have configured the voicemail.conf file as per the wiki to email voicemails as an attachment. I cannot find any instructions/locations to set the outgoing server login information. Furthermore, I can get no emails from asterisk. Can anyone point me to the next step to setup the attachment of voicemail messages to an email? Thanks BEN ___ --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
RE: [Asterisk-Users] Middle Ground between POTS and T1?
Here (Croatia) is also possible to get partial E1 (PRI/PRA)... from one of our telcos (DT T-com) we can get PRA in 10 increments: 10B, 20B and 30B We have a partial T1 (5B + D, iirc) from Allstream - there may be a provider in your area that does something similar. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 17-Oct-05, at 12:48 PM, Matthew T. O'Connor wrote: I was wondering if there was a middle ground between POTS lines and a T1. I have a new office with a T1 line and while it's working well, it's a lot of money and we will never use anywhere near 23 lines at one time. Is it possible to get a few ISDN lines or something and bundle them together? Basically I would like to get the digital features of the T1 PRI (DID number, etc...) but smaller. Thanks, Matthew ___ --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] sip show peers
They do not have NAT option.. and they do not have qualify... Hi, I have 3 SIP extensions, setup as follows: # Device Location options 200 Sipura local 210 Sipura remote nat=yes qualify=yes 310 eyebeam remote nat=yes qualify=yes This is the result of sip show peers: Name/user Host Dyn Nat Status 200/200 192.168.1.150 D Unmonitored 210/210 84.36.36.140 D N OK (305 ms) 310/310 71.180.126.60 D N Unmonitored Does anyone know why ext 210 the only one has a ping status OK (305 ms) and the others are Unmonitored Regards ___ --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] SS7 with Asterisk
Name of the company is MULTI-line GmbH You can contact mr. Zlatko Medibach at +43 1 78932320 or on cellular +43 676 3220262. Mail: [EMAIL PROTECTED] Their HQ is in Wien.. I can not help you with the details, I just know that they implemented SS7 on * for some telcos there. Goran -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Johann Steinwendtner Sent: Tuesday, October 11, 2005 1:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SS7 with Asterisk Goran, which company ist this ? Do they use the www.ss7box.com approach ? Thanks and best regards Hans Goran Skular schrieb: anyone running SS7 with Asterisk ? Please help me out. I need to know the hardware used for SS7 with Digium E1 cards... I can point you to one company in Austria. They deployed SS7 on Asterisk, but not with Digium cards for one smaller telco. ___ --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
[Asterisk-Users] arcaplex / horizon isdn and analog multiplex
Has anybody tried something like this: http://www.arca-technologies.com/datasheets/arcaplexhorizon.pdf It will be interesting to have ability to make systems like: SCENARIO 1 (2 incoming BRI lines and 12 analog extensions with ability to connect additional isdn devices to s0 buses): 1 card with 8 BRI (from Junghanns or Beronet or someone else) ports (2 of them configured in TE mode and 6 of them in NT mode) 1 something that will convert e.g. 6 BRI to 12 analog FXS ports for analog telephony equipment.. Or SCENARIO 2 (1 incoming PRI E1 or T1 and 32 or more analog extensions) 2 Digium/Sangoma/Eicon whatsoever T1 or E1 cards (1 to telco, 1 to something like this arcaplexhorizon) 1 Arcaplexhorizon ISDN and analog multiplexor with 32 analog ports (PO21/32A) Maybe this Arcaplex can be used for 32 analog ports connected to Asterisk with 1E1/T1 card Some thoutghts ? Goran ___ --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] Outgoing call: hangup after answer
For your information.. if someone get in the same trouble.. problem is solved, but not with the software We just changed our BRI NT device with a different one.. from now on it works very well We had Elcon NT1+2a/b and now it is replaced with Santis ISDN NT1+2ab Here is pri debug: -- Making new call for cr 143 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=22 Call Ref: len= 1 (reference 15/0xF) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 01 81] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Preferred Dchan: 0 ChanSel: B1 channel ] [6c 05 21 80 32 30 30] Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '200' ] [70 01 c1] Called Number (len= 3) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '' ] -- Called g1/ Protocol Discriminator: Q.931 (8) len=11 Call Ref: len= 1 (reference 143/0x8F) (Terminator) Message type: SETUP ACKNOWLEDGE (13) [18 01 89] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference 15/0xF) (Originator) Message type: INFORMATION (123) [70 02 c1 30] Called Number (len= 4) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0' ] Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference 15/0xF) (Originator) Message type: INFORMATION (123) [70 02 c1 39] Called Number (len= 4) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0' ] Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 143/0x8F) (Terminator) Message type: CALL PROCEEDING (2) -- Zap/1-1 is making progress passing it to SIP/200-7b76 Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference 143/0x8F) (Terminator) Message type: ALERTING (1) [1e 02 84 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 30 (cs0, Progress Indicator) n Zap/1-1 is ringing NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Delivered, peerstate Call Received Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference 15/0xF) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' Hi, When we make an outgoing call on ISDN (zaphfc) with overlap dialing we get immidiate hangup after answer. But when we place a full number before dialing everything is ok. Any help appriciated!! Thanks here is info with debug: [1e 02 84 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] [1e 02 84 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 30 (cs0, Progress Indicator) -- Zap/1-1 is ringing, hanging up. NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Delivered, peerstate Call Received Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference 64/0x40) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] ___ --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:
RE: [Asterisk-Users] Open Source Content Management System - Joomla
Our Web is based on Mambo portal software and it is connected with our Asterisk installation. We wrote our own CDR rating engine and modules for Mambo. Also, you can register for VoIP termination services inside mambo.. we wrote one component and couple of modules. So, when user create an account on mambo, asterisk account is also created automaticly (if choosed... cron script every 10 minutes) ... CDR rating is simple, but it works.. there is no fancy things in rating like tariffs or similar... (at this moment) You can check on www.slsolucije.hr .. it is on Croatian.. but.. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Monday, October 10, 2005 11:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Open Source Content Management System - Joomla And how exactly is Asterisk relevant to a CMS? could you give a more specific example? This is relevant where Administrative users wanted to manage their Asterisk GUI setups like [EMAIL PROTECTED], AMP, Phonecall etc Seshu NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --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
RE: [Asterisk-Users] SS7 with Asterisk
anyone running SS7 with Asterisk ? Please help me out. I need to know the hardware used for SS7 with Digium E1 cards... I can point you to one company in Austria. They deployed SS7 on Asterisk, but not with Digium cards for one smaller telco. ___ --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] Outgoing quality
We were using ilbc at first.. but it shows that it really needs a lot of time for transcoding.. (CLI: show translation) resulting with hearable delays. So, where bandwidth is an issue, try to use g729. We are also using gsm, it seems that it works very well. OK .. what about ilbc ... could it be a decent choice?? -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] conto di Mojo with Horan Company, LLC Inviato: lunedì 10 ottobre 2005 22.00 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Outgoing quality Are you calling from a soft- or hardphone on a network with a high amount of latency? If your (for example) SIP phone can't deliver voice packets to asterisk in time for asterisk to put them where they belong in the Zap channel, things like this might happen. Usually the interruptions could be described as clicks or crackles. In this case, you could reduce the network traffic by utilizing a codec with a smaller bandwidth usage, like g729 or gsm if your phone supports it. Fabrizio Mazzoni wrote: ___ --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] [Fwd: Libpri/chan_zap problems?]
What am I doing wrong here? Why is this happening? libpri is version 1.0.7-1 (debian package) asterisk is version 1.0.7.dfsg.1-2 (debian package) zaptel is version 1.0.9.2 -- Executing Dial(SIP/739-5935, Zap/g1/0916000739) in new stack -- Called g1/0916000739 -- Channel 0/1, span 1 got hangup Oct 10 13:14:45 WARNING[7544]: app_dial.c:412 wait_for_answer: Unable to forward voice Does the same thing happens even when you're not calling cellular number VIP (I assume you are in Croatia, calling VIPnet) i.e. some fixed line number ? And what connection do you use, BRI (bristuff, capi), PRI, some FXO,... You can reach me at 01/4573573. I'll be glad to hear you if my assumptions (on Croatia thing) were right... ___ --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] Outgoing call: hangup after answer
Hi,
When we make an
outgoing call on ISDN (zaphfc) with overlap dialing we get immidiate hangup
after answer. But when we place a full number before dialing everything is ok.
Any help appriciated!! Thanks
here is info with
debug:
== Primary
D-Channel on span 1 up -- Executing Dial("SIP/200-164c",
"zap/g1/|100|tc") in new stack-- Making new call for cr
192 -- Requested transfer capability: 0x00 -
SPEECH Protocol Discriminator: Q.931 (8) len=22 Call Ref:
len= 1 (reference 64/0x40) (Originator) Message type: SETUP (5)
[04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std:
0 Info transfer capability: Speech
(0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Ext: 1 User information layer 1: A-Law (35) [18 01 81]
Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Preferred
Dchan:
0
ChanSel: B1
channel
] [6c 05 21 80 32 30 30] Calling Number (len= 7) [ Ext: 0
TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163)
(1)
Presentation: Presentation permitted, user number not screened (0) '200'
] [70 01 c1] Called Number (len= 3) [ Ext: 1 TON:
Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
'' ] -- Called g1/ Protocol Discriminator: Q.931
(8) len=11 Call Ref: len= 1 (reference 192/0xC0)
(Terminator) Message type: SETUP ACKNOWLEDGE (13) [18 01
89] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0,
Exclusive Dchan:
0
ChanSel: B1
channel
] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1
Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network
serving the local user
(2)
Ext: 1 Progress Description: Inband information or appropriate pattern now
available. (8) ]-- Processing IE 24 (cs0, Channel Identification)--
Processing IE 30 (cs0, Progress Indicator) Protocol Discriminator: Q.931
(8) len=8 Call Ref: len= 1 (reference 64/0x40)
(Originator) Message type: INFORMATION (123) [70 02 c1
39] Called Number (len= 4) [ Ext: 1 TON: Subscriber Number
(4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '9' ]
Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1
(reference 64/0x40) (Originator) Message type: INFORMATION (123)
[70 02 c1 35] Called Number (len= 4) [ Ext: 1 TON: Subscriber
Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5'
] Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1
(reference 192/0xC0) (Terminator) Message type: CALL PROCEEDING
(2) -- Zap/1-1 is making progress passing it to
SIP/200-164c Protocol Discriminator: Q.931 (8) len=12 Call
Ref: len= 1 (reference 192/0xC0) (Terminator) Message type: ALERTING
(1) [1e 02 84 88] Progress Indicator (len= 4) [ Ext: 1
Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network
serving the remote user
(4)
Ext: 1 Progress Description: Inband information or appropriate pattern now
available. (8) ] [1e 02 84 82] Progress Indicator (len= 4) [
Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public
network serving the remote user
(4)
Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ]--
Processing IE 30 (cs0, Progress Indicator)-- Processing IE 30 (cs0, Progress
Indicator)
-- Zap/1-1 is ringing, hanging
up.
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Delivered,
peerstate Call Received Protocol Discriminator: Q.931 (8)
len=8 Call Ref: len= 1 (reference 64/0x40) (Originator) Message
type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext:
1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private
network serving the local user
(1)
Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1)
] -- Hungup 'Zap/1-1' Protocol Discriminator:
Q.931 (8) len=4 Call Ref: len= 1 (reference 192/0xC0)
(Terminator) Message type: RELEASE (77)NEW_HANGUP DEBUG: Calling
q931_hangup, ourstate Null, peerstate Release Request Protocol
Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference
64/0x40) (Originator) Message type: RELEASE COMPLETE (90) [08 02
81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0:
0 Location: Private network serving the local user
(1)
Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1)
]NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate
NullNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate
Null -- Executing Macro("SIP/200-164c", "hangupcall") in
new stack -- Executing ResetCDR("SIP/200-164c", "w") in
new stack -- Executing NoCDR("SIP/200-164c", "") in new
stack -- Executing Wait("SIP/200-164c", "5") in new
stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/200-164c' in macro 'hangupcall' == Spawn extension
(from-internal, h, 1) exited non-zero on
'SIP/200-164c'
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[Asterisk-Users] overlap zaphfc - dialtone
Hello all, I have a problem with overlap dialing and don't know how to get rid of it. My setup is: 1 HFC card with bristuff - ZAP/g1 (2B + 1D channels), SIP phones (I just removed TDM400P with 4 FXS) I created test extension 222 which goes directly to g1. In Zapata.conf overlapdial is set to yes. First I created this extension: exten = 222,1,Dial(zap/g1,100,tc) and channel got hangup every time. So I even saw bug 4913 http://bugs2.digium.com/view.php?id=4913nbn=1 and bug 4771. But that wasn't my problem my problem is that I didn't included / after g1.. So, I changed that and now my extension look like: exten = 222,1,Dial(zap/g1/,100,tc) This solved the problem with line being hangup-ed like in bug 4913, and I am getting the telco dialtone. So, when dialing 222 I get: A12*CLI -- Executing Dial(SIP/200-dd52, zap/g1/|100|tc) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/ After that, I can even dial from this dialtone, but when called party rings I get the following message and auto hangup: -- Zap/1-1 is making progress passing it to SIP/200-dd52 -- Zap/1-1 is ringing, hanging up. -- Hungup 'Zap/1-1' The called phone stops ringing, Zap channel hangs up, And SIP phone is still on the air without anybody. Thank You all for help, Goran ___ --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] overlap zaphfc - dialtone
My setup is: 1 HFC card with bristuff - ZAP/g1 (2B + 1D channels), SIP phones (I just removed TDM400P with 4 FXS) I created test extension 222 which goes directly to g1. In Zapata.conf overlapdial is set to yes. First I created this extension: exten = 222,1,Dial(zap/g1,100,tc) and channel got hangup every time.. So I even saw bug 4913 http://bugs2.digium.com/view.php?id=4913nbn=1 and bug 4771.. But that wasn't my problem. my problem is that I didn't included / after g1.. So, I changed that and now my extension look like: exten = 222,1,Dial(zap/g1/,100,tc) This solved the problem with line being hangup-ed like in bug 4913, and I am getting the telco dialtone. So, when dialing 222 I get: -- Executing Dial(SIP/200-dd52, zap/g1/|100|tc) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/ -- Zap/1-1 is making progress passing it to SIP/200-dd52 -- Zap/1-1 is ringing, hanging up. -- Hungup 'Zap/1-1' It seems that when it detects Ringing, Asterisk executes Hangup in the same time But why? ___ --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
