[asterisk-users] What is ZapRas used for ?
Hi list, What is ZapRas used for ? I would like to use asterisk as a RAS server replacing a Cisco RAS server where users calls to a number directed to asterisk, and here, asterisk answer the data calls and assign an IP address via PPP to calling user. Is is possible ? Thanks. ___ --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] SMS over SIP and Asterisk ??
Well, I mean Instant messaging between two SIP users registered on Asterisk-sip server. The thing is, some sip phones supports instant messaging but, how can I get this feature work in asterisk ? Angel Date: Wed, 13 Jul 2005 20:17:22 -0400 From: Shidan [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SMS over SIP and Asterisk ?? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Do you mean SMS or a a SIP MESSAGE, the only sure way I can think of to send SMS with SIP, and I'm no SIP expert here, is if there was an SMS MIME type and you just used SIP for the transport, and even if there is such a type I doubt anyone has implemented anything for it yet, let alone *. As to does Asterisk support MESSAGE requests with a plain/text MIME type, you can use the ap. SendText() , look it up on the wiki Regards, Shidan http://www.nuovotel.com On 7/13/05, Angel Diaz [EMAIL PROTECTED] wrote: Hi, Is there a way to send and receive SMS over SIP protocol with Asterisk ? I mean, between two SIP phones like below... SIP_phone A (sending sms) Asterisk SIP_phone B (receiving sms) ...Is it possible ? If so, how could I do it ? Thanks, Angel. ___ 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] SMS over SIP and Asterisk ??
Hi, Is there a way to send and receiveSMS overSIP protocolwith Asterisk ? I mean, between two SIP phones like below... SIP_phone "A"(sending sms) Asterisk SIP_phone "B" (receiving sms) ... Is it possible ? If so, how could I do it ? Thanks, Angel. ___ 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] VoIP services
Hi guys, Can somebody help me on some questions please ? If I have a VoIP network with my Asterisk platform in Europe, what do I need to interconnect my VoIP network to another network in the USA in order to my customers in Europe be able to call to customers in the USA network ? The network in the USA is not an Asterisk platform. Thanks, ___ 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] Connect 30 phone lines to asterisk how to
Hi, I have to connect 30 phone lines to my asterisk server, can somebody help on how I have to do it ? I have a TDM405P and one TDM400P with 4 FXO ports. Do I have to use 8 TDM400P ? Or, is there another way to do it ? Thanks, Angel. ___ 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] Play Sound File Without Answer Channel ??
Hi list, Is there anyway to play a Sound File without answering the channel ? I have tried Playback(myfile, noanwer), but there is no audio there... Could you help me please ? Angel ___ 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] Play Sound File Without Answer Channel
Mikael, Well, to be more specific, I'm using ISDN PRI. 30B+D. - Original Message - From: Angel Diaz [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, April 11, 2005 3:55 PM Subject: Re: [Asterisk-Users] Play Sound File Without Answer Channel I'm using Zap channels. Does Zap channels support ? Thanks, Ange Date: Mon, 11 Apr 2005 20:35:58 +0200 From: Mikael Magnusson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Play Sound File Without Answer Channel ?? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed Angel Diaz wrote: Hi list, Is there anyway to play a Sound File without answering the channel ? I have tried Playback(myfile, noanwer), but there is no audio there... Could you help me please ? Angel It's working with an ISDN phone and zaphfc and asterisk with bristuff patches for me. What type of channel are you using? According to show application playback: Not all channels support playing messages while still hook. /Mikael Magnusson ___ 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] Re: Asterisk-Users Digest, Vol 9, Issue 96
I'm using Zap channels. Does Zap channels support ? Thanks, Ange Date: Mon, 11 Apr 2005 20:35:58 +0200 From: Mikael Magnusson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Play Sound File Without Answer Channel ?? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed Angel Diaz wrote: Hi list, Is there anyway to play a Sound File without answering the channel ? I have tried Playback(myfile, noanwer), but there is no audio there... Could you help me please ? Angel It's working with an ISDN phone and zaphfc and asterisk with bristuff patches for me. What type of channel are you using? According to show application playback: Not all channels support playing messages while still hook. /Mikael Magnusson ___ 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] Play Sound File Without Answer Channel
I'm using Zap channels. Does Zap channels support ? Thanks, Ange Date: Mon, 11 Apr 2005 20:35:58 +0200 From: Mikael Magnusson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Play Sound File Without Answer Channel ?? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed Angel Diaz wrote: Hi list, Is there anyway to play a Sound File without answering the channel ? I have tried Playback(myfile, noanwer), but there is no audio there... Could you help me please ? Angel It's working with an ISDN phone and zaphfc and asterisk with bristuff patches for me. What type of channel are you using? According to show application playback: Not all channels support playing messages while still hook. /Mikael Magnusson ___ 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] Play Sound File Without Answer Channel
I want to use the Voicemail app and before that, I would like to play an audio file but not billable in the Switch side. Than, to do so, I have to be able to no send the Answer message during the play of the file. Then after finish the file, I'w xecute the Voicemail app. That's why I need to play the file before answer the channel. Is it possible ? I have looked at the Playback and Background app, and I see they are answering the channel before playing the file. Angel. Date: Mon, 11 Apr 2005 15:40:59 -0500 From: Eric Wieling aka ManxPower [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Play Sound File Without Answer Channel To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed Angel Diaz wrote: Mikael, Well, to be more specific, I'm using ISDN PRI. 30B+D. - Original Message - From: Angel Diaz [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, April 11, 2005 3:55 PM Subject: Re: [Asterisk-Users] Play Sound File Without Answer Channel I'm using Zap channels. Does Zap channels support ? Yes, but it would depend on your provider. -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest, Vol 9, Issue 97 * ___ 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 - SS7 or ISDN
I have the same problem :( Steve, may we have some specifications about your SS7 - ISUP solution ? Angel. Message: 16 Date: Wed, 06 Apr 2005 02:28:13 -0500 From: Brian Capouch [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk - SS7 or ISDN To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed Roy Sigurd Karlsbakk wrote: 1.Does Asterisk support SS7 and ISDN? ISDN is supported out of the box. SS7 support is (or will soon be?) supported by a commercial version of Asterisk. Search the list archives or post to asterisk-biz. Steve Underwood (here on the list) has a commercial ss7 solution for asterisk. Does anyone know how to find out any of the specs on the product, particularly what it costs to license? I have sent him a small river of mails over the past six or eight months asking that question, which seems pretty primal. I've never gotten a response. I suppose now I'll hear that my mail agent is eating his responses. That would actually be *good* news. Thanks. B. ___ 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 2.0 Stable release out now
This version support SS7 - ISUP protocol ? Does some body know where can I find it ? I mean, Asterisk SS7... Angel. Message: 21 Date: Fri, 01 Apr 2005 09:40:33 +0200 From: Olle E. Johansson [EMAIL PROTECTED] Subject: [Asterisk-Users] *** Asterisk 2.0 Stable release out now To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed During the developer's conference call yesterday evening, it was decided that we finally should release the much-awaited Asterisk 2.0 Stable release, also called codename AAFJ. This relaese is based on the hidden cvs that has been in operation for six months by a group of core development members in the Asterisk.org open source project, under the leadership of Brian K. East, who will maintain the stable code base for the 2.0 CVS tree and releases. -It's awsome, says Brian, but the new features I'm adding to 2.0.1 stable will be even more spectacular. Follow me to the future! Among the new features in Asterisk 2.0 is * APBX - A fully pluggable PBX architecture - The APBX framework makes everything in Asterisk 2.0 hot-pluggable and dynamic, including the PBX itself. With this framework, Asterisk 2.0 will be able to be the host system for almost anything, including the famous Apache.org web server, the SipFoundry SIPx PBX and a Java Runtime Engine. Rumours has it that one developer actually ported the Erlang runtime and executed an Ericsson AXE switch within Asterisk. With an embedded web server, we can finally start working on a decent user interface model says Kram Spencer, the original developer of Asterisk. * DBRAGI - The Database Remote procedure call AGI subsystem -- The DBRAGI subsystem makes it possible to move the dial plan processing to stored procedures in databases. With Asterisk 1.2, the ARA (Asterisk Realtime Architecture) took a first step towards a better database integration. With 2.0, the project actually runs most of the PBX within an Oracle (TM) database, making Asterisk carrier grade. * XIAX - The New Inter-Asterisk Protocol -- With Asterisk 2.0, the project also launches the next generation of the IAX protocol. This is a huge update of the rather oldfashioned IAX protocol engine. - XML based messages All messages in XIAX is based on XML. This makes the protocol more robust, since all messages are checked for correct syntax with an external DTD and XML parser. All voice frames are encoded in BASE64 and checked with an S/MIME signature, which makes the XIAX protocol the most secure VoIP protocol in the known universe. - Full DNS NAPTR/SRV support To add to the robustness of the protocol, all communication is done with full DNS service names. For each packet in the data stream, there's full redundancy based on DNS lookups. The recommendation for XIAX is to define at least five XIAX servers per phone number, and let DNS route the XIAX packets. No packet will get lost, due to the stability and simpleness of the DNS system. says Kram. Using IP numbers did not gives us this functionality. - Strong TCP/SSL support The new XIAX protocol also supports TCP with SSL encapsulation. TCP is much easier for the firewall to handle and with strong SSL encryption. With IAX2 we could bypass every NAT device. With XIAX over SSL on the HTTP port, we can traverse any firewall too. says Steve Xintaro, the main architect of XIAX. * New source code structure - C# and .net Asterisk 2.0 was moved to a Microsoft platform due to the demand for higher stability and a more secure foundation. Therefore, the code was quickly moved to C# on the .net platform. This gives Asterisk a lot of new features, including being fully integrated with Microsoft Exchange and Microsoft Active Directory. With all the user data stored in Active Directory, we finally have the user under full control. Users can dial in to the PBX to change their Windows password. We can also implement single-sign-on based on DTMF from a cell phone or WiFi phone. says Kelvin Reming. The C# language gives us much more modern code. And I'm so happy to get rid of the stupid-looking arctic bird, an ugly animal that that couldn't even fly. * New user-support system: SmartyList (TM) In order to solve the problem with the asterisk-users mailing list that was the main support channel for old Asterisk versions, the Asterisk 2 team also constructed the SmartyList auto-support system, that will automatically analyze all input and sort it out on one of twenty different lists. Eighteen of these are automatically handled by auto-responders, that
[Asterisk-Users] Voicemail Question...help
Hi all, Can anybody help me on this ? I'm trying to use asterisk voicemail application. The thing is, when a call forward from the switch arrive to asterisk over ISDN PRI I verify if the redirecting number match one subscriber of the asterisk voicemail databases otherwise, forward this call to another voicemail platform. My question is, how I have to forward this call to the other voicemail platform via ISDN PRI ?? (take into account I have to send the redirecting number to the other platform). I was thinking it could be via Dial() app, but I think it will remove the redirecting number here This is the scenario. Switch - Asterisk Voicemail(here, chek is the user is on the database) , otherwise send the call to - VMS(other). ___ 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] VMS - AGI
Hi list, I would like to use the * VMS application with a GSM network, I know that * support Unavailable and Busy Redirecting Reason in the extensions.conf but, what's about the No Reply ?? Then, I know that doing a debug on the PRI, I see the redirecting reason... My question is; is there anyway to obtain this redirecting reason through perl AGI ?? Thanks in advance . Angel. ___ 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] Swap Memory get used totally
Hi list, Time to time, my asterisk goes down.Verifying with TOP, I see the swap memory of the computer get used totally but, I don't see what the process is using it. Hereis a copy wath I see doing top. Does somebody have an idea ? My asterisk version is Asterisk CVS-HEAD-08/18/04-22:30:24 Thanks Angel. 08:49:19 up 5:23, 1 user, load average: 0.50, 0.70, 0.64 35 processes: 33 sleeping, 2 running, 0 zombie, 0 stopped CPU states: 19.4% user 11.2% system 0.0% nice 0.0% iowait 69.4% idle Mem: 222992k av, 191988k used, 31004k free, 0k shrd, 68604k buff 120700k actv, 464k in_d,2384k in_c Swap: 457844k av,1060k used, 456784k free 67764k cached PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME CPU COMMAND 1238 root 15 0 6280 6212 1360 S10.5 2.7 3:49 0 asterisk 31788 root 16 0 1456 1456 1100 S 0.0 0.6 0:00 0 bash 1200 mysql 16 0 1424 1424 896 S 0.0 0.6 0:02 0 mysqld 31855 root 15 0 1180 1180 860 R 0.0 0.5 0:00 0 top 644 admin 15 0 948 948 600 S 0.0 0.4 0:00 0 bftpd 1138 root 15 0 616 524 372 S 0.0 0.2 0:00 0 sshd 1222 daemon15 0 176 176 120 S 0.0 0.0 0:00 0 atd 1235 root 15 0 168 16896 S 0.0 0.0 0:00 0 bftpd 992 root 15 0 188 156 112 S 0.0 0.0 0:00 0 syslogd 1164 root 25 0 152 152 0 S 0.0 0.0 0:00 0 safe_mysqld 1181 root 15 0 152 15288 S 0.0 0.0 0:00 0 crond 1229 root 25 0 136 136 0 S 0.0 0.0 0:00 0 S99local 1 root 15 0 112 8456 S 0.0 0.0 0:04 0 init 702 root 25 0 2884 0 S 0.0 0.0 0:00 0 rc 996 root 15 0524 0 S 0.0 0.0 0:00 0 klogd 1100 root 25 0524 0 S 0.0 0.0 0:00 0 apmd 1152 root 24 0 1244 0 S 0.0 0.0 0:00 0 xinetd 2 root 15 0 00 0 SW0.0 0.0 0:00 0 keventd 3 root 15 0 00 0 SW0.0 0.0 0:00 0 kapmd 4 root 34 19 00 0 SWN 0.0 0.0 0:01 0 ksoftirqd_CPU0 9 root 25 0 00 0 SW0.0 0.0 0:00 0 bdflush 5 root 15 0 00 0 SW0.0 0.0 0:01 0 kswapd 6 root 15 0 00 0 SW0.0 0.0 0:00 0 kscand/DMA 7 root 15 0 00 0 SW0.0 0.0 0:02 0 kscand/Normal 8 root 15 0 00 0 SW0.0 0.0 0:00 0 kscand/HighMem 10 root 15 0 00 0 SW0.0 0.0 0:00 0 kupdated 11 root 23 0 00 0 SW0.0 0.0 0:00 0 mdrecoveryd 15 root 15 0 00 0 SW0.0 0.0 0:01 0 kjournald 617 root 15 0 00 0 SW0.0 0.0 0:00 0 kjournald 942 root 15 0 00 0 SW0.0 0.0 0:00 0 eth1 1014 rpc 23 0760 0 SW0.0 0.0 0:00 0 portmap 1033 rpcuser 25 0800 0 SW0.0 0.0 0:00 0 rpc.statd ___ 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] Re: Asterisk-Users Digest, Vol 4, Issue 222
Hi Steve, It will support ISUP v2 or it will be MAP or just MTP1, MTP2,MTP3 ? I'm very interested in ISUP for the moment... but if it will support MAP as well .. great ! Thanks, Angel. -- Message: 5 Date: Wed, 17 Nov 2004 09:05:21 +0800 From: Steve Underwood [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SS7 for * To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi Angel, It is working pretty well. I think it will be available about the end of the year. I will not be free. It will be supplied with a commercially licenced Asterisk. Regards, Steve Angel Diaz wrote: Hi all, Does somebody know what's new with SS7 and * ? I'm very interested. Is it ready ? I'm prepared to pay if necessary. Thanks, Angel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SS7 for *
Hi all, Does somebody know what's new with SS7 and * ? I'm very interested. Is it ready ? I'm prepared to pay if necessary. Thanks, Angel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re:Re: [Asterisk-Users] Fax and Asterisk
Hi Matt Riddell Here you have the write permission for all drwxrwxrwx2 root root 4096 Sep 14 16:09 incoming And about your question; How long does it time out for? It stay there without hang up until I switch off the fax machine. I have installed tiff-v3.6.0 and spandsp-0.0.1k.tar.gz and performed all steps described by Steve Underwood. Thanks, Angel. Message: 6 Date: Fri, 17 Sep 2004 12:29:10 +1200 From: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fax and Asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII On 15 Sep 2004 at 15:10, Angel Diaz wrote: Hi all, I have problems with rxfax application. It seems to be ok but I don't receive the fax in my directory. My extension.conf is as follow: -- SNIP -- And my log is : -- Executing Answer(Zap/1-1, ) in new stack -- Executing BackGround(Zap/1-1, 00) in new stack -- Playing '00' (language 'es') -- Redirecting Zap/1-1 to fax extension == Spawn extension (reception, fax, 0) exited non-zero on 'Zap/1-1' -- Executing Goto(Zap/1-1, fax1|100|1) in new stack -- Goto (fax1,100,1) -- Executing Macro(Zap/1-1, fax) in new stack -- Executing SetVar(Zap/1-1, FAXFILE=/var/spool/asterisk/incoming/1095285200.16.tif) in new stack -- Executing RxFAX(Zap/1-1, /var/spool/asterisk/incoming/1095285200.16.tif) in new stack and after that, it stay there without time out. Any suggestion ? It's probably receiving the fax. How long does it time out for? Who can write to that directory (assuming you've created /var/spool/asterisk/incoming/)? Cheers, Matt Riddell (New Zealand Digium Distribution/Custom Software) http://www.sineapps.com/contact.php (contact us) http://www.sineapps.com/news.php (asterisk news) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 2, Issue 163
Hi Matt, I have verified with ztmonitor the audio level and it was too low, then with this the fax machine report Not Response. I modified the audio level in zapata.conf and after that the fax machine report Commnunication Error. Do you an idea what could be ? Thanks, Angel. Message: 3 Date: Sat, 18 Sep 2004 00:48:23 +1200 From: [EMAIL PROTECTED] Subject: Re:Re: [Asterisk-Users] Fax and Asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII On 17 Sep 2004 at 8:29, Angel Diaz wrote: Hi Matt Riddell Here you have the write permission for all drwxrwxrwx2 root root 4096 Sep 14 16:09 incoming And about your question; How long does it time out for? It stay there without hang up until I switch off the fax machine. I have installed tiff-v3.6.0 and spandsp-0.0.1k.tar.gz and performed all steps described by Steve Underwood. Thanks, Angel. LOL the permissions shouldn't be a problem. Redirecting Zap/1-1 to fax extension Isn't this line supposed to say detected fax, redirected to fax extension? Are you doing anything weird in Extensions.conf? Also check the volume of the incoming audio by typing: /usr/src/zaptel/ztmonitor x -v (where x is the incoming channel i.e 1 or 2 or 3 etc) The bar should go pretty close to the right hand side of the line, but shouldn't sit at the top. How's it look for you? Matt Riddell (New Zealand Digium Distribution/Custom Software) http://www.sineapps.com/contact.php (contact us) http://www.sineapps.com/news.php (daily asterisk news) Message: 6 Date: Fri, 17 Sep 2004 12:29:10 +1200 From: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fax and Asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII On 15 Sep 2004 at 15:10, Angel Diaz wrote: Hi all, I have problems with rxfax application. It seems to be ok but I don't receive the fax in my directory. My extension.conf is as follow: -- SNIP -- And my log is : -- Executing Answer(Zap/1-1, ) in new stack -- Executing BackGround(Zap/1-1, 00) in new stack -- Playing '00' (language 'es') -- Redirecting Zap/1-1 to fax extension == Spawn extension (reception, fax, 0) exited non-zero on 'Zap/1-1' -- Executing Goto(Zap/1-1, fax1|100|1) in new stack -- Goto (fax1,100,1) -- Executing Macro(Zap/1-1, fax) in new stack -- Executing SetVar(Zap/1-1, FAXFILE=/var/spool/asterisk/incoming/1095285200.16.tif) in new stack -- Executing RxFAX(Zap/1-1, /var/spool/asterisk/incoming/1095285200.16.tif) in new stack and after that, it stay there without time out. Any suggestion ? It's probably receiving the fax. How long does it time out for? Who can write to that directory (assuming you've created /var/spool/asterisk/incoming/)? Cheers, Matt Riddell ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax and Asterisk
Hi all,
I have problems with rxfax application. It seems to be ok but I don't receive the fax in my directory.
My extension.conf is as follow:
[macro-fax]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/incoming/${UNIQUEID}.tif)exten = s,2,rxfax(${FAXFILE})
[fax]
exten = 100,1,macro(fax)
[reception]exten =s,1,Answer()exten =s,2,Background(00)
exten =fax,1,Goto(fax,100,1)
And my log is :
-- Executing Answer("Zap/1-1", "") in new stack
-- Executing BackGround("Zap/1-1", "00") in new stack
-- Playing '00' (language 'es')
-- Redirecting Zap/1-1 to fax extension
== Spawn extension (reception, fax, 0) exited non-zero on 'Zap/1-1'
-- Executing Goto("Zap/1-1", "fax1|100|1") in new stack
-- Goto (fax1,100,1)
-- Executing Macro("Zap/1-1", "fax") in new stack
-- Executing SetVar("Zap/1-1", "FAXFILE=/var/spool/asterisk/incoming/1095285200.16.tif") in new stack
-- Executing RxFAX("Zap/1-1", "/var/spool/asterisk/incoming/1095285200.16.tif") in new stack
and after that, it stay there without time out. Any suggestion ?
Thanks,
Angel.
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[Asterisk-Users] Is posissble TE405P ?
Hi all: Is it possible to setup the TE405P as follow for PRI E1 ? zaptel span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 and the second span as well as span=2,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 The thing is, I am trying to connect * to Nortel DMS100 MSC which does not support the way to configure ISDN cards in sequential channel mode like 1-15, 17 -31 and 32-46, 48-62... Can somebody help me ? Thanks, Angel, Do you Yahoo!? Win 1 of 4,000 free domain names from Yahoo! Enter now.
[Asterisk-Users] Question about TE405P
Dear all: Doesanybody know is it possible to use the board TE405P with only one port configured as follow, or I have to use the 4 ports at the same time ? Thanks, Angel ZAPTEL span=1,1,0,ccs,hdb3bchan=1-15dchan=16bchan=17-31 ZAPATA [channels]context=menu-general switchtype=euroisdn pridialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=noechocancel=yes immediate=nochannel = 1-15,17-29__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
[Asterisk-Users] Question about TE405P
Hi all, Does somebody know how I have to setup my TE405P ? Is it correct my configuration below ? Otherwise, can somebody help me ? Thanks, Angel. zaptel.conf span=1,1,0,ccs,hdb3 span=2,0,1,ccs,hdb3 span=3,0,1,ccs,hdb3 span=4,0,1,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 zapata.conf [channels] context=default switchtype=euroisdn pridialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no callwaitingcallerid=yes language=en immediate=no channel = 1-15,17-31 channel = 1-15,17-31 channel = 1-15,17-31 channel = 1-15,17-31 Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers!
[Asterisk-Users] Connect 16 E1/T1 between * and other switch...
Hi all, I'm looking the way to connect 16 E1/T1 to my * server. How it would be possible ? . Does anyone have already this experience ? I'm thinking to do using two * machine servers. One card E400XX or T400XX from digium on each machine and interconnect one each other using IAX. One * as a master, and the other which will connect to this (master) to use its numbering plan... I am correct ? Thanks in advance .. Angel Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard.
[Asterisk-Users] Problems with Zaptel
Hi all: I have problems to setup my zaptel E100P hardware. When I start * after receive the "Asterisk Ready" I see this: *CLI Jun 22 20:37:55 NOTICE[1133718080]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 1 Jun 22 20:37:55 NOTICE[1133718080]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 2 Up to channel 31. anfter this: Jun 22 20:37:55 WARNING[1125329600]: chan_zap.c:5942 zt_pri_error: PRI: Read on 39 failed: Unknown error 500 PRI got event: 5 Jun 22 20:37:59 WARNING[1125329600]: chan_zap.c:5942 zt_pri_error: PRI: Read on 39 failed: Unknown error 500PRI got event: 8 My [zaptel.conf ] is as below: span=1,1,0,ccs,hdb3bchan=1-15bchan=17-31dchan=16 My [zapata.conf] is as below: [channels] context=incoming-default switchtype=national signalling=pri_net usecallerid=yes hidecallerid=no callwaitingcallerid=yes echocancel=yes immediate=no channel = 1-15 channel = 17-31 What is wrong ? Could you help me please ? thanks. Angel__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
[Asterisk-Users] Re: No B-Channels. PRI. E100P. HELP!
Hi -A Just an information about E1s. The E1 use TS (time slot) from 0 to 31. TS 0 is for the synchronization, TS 1 to 15 are for Speech (B channels), TS 16 (D channel) is for the signalling, TS 17 to 31 are for Speech (B channels). I hope this help you, Angel. On Tuesday 15 June 2004 03:17, Holger Schurig wrote: bchan=1-15 dchan=16 bchan=17-31 Just a wild guess (I never worked with this equipment): try bchan=1-15,17-31 dchan = 16 By the way: what is on channel 0 ?E1s start at channel 0?I don't know anything about E1s, but with T1 PRI the D channel is usually the last channel (channel 24 for PRI). I imagine he's correct since the telco says everything's cool with the d channel on 16, but I figured I'd throw the question out anyway. :-)-A. Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses.
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #4101 - 12 msgs
Thanks Andy ! that is I was looking for. It works
fine.
Angel.
Date: Fri, 11 Jun 2004 00:55:04 +0200
From: Andy Powell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] How to get the Called
id with AGI
Reply-To: [EMAIL PROTECTED]
On 10/06/2004 at 14:40 Angel Diaz wrote:
Hi all,
Is there a way to get the called id (the B
number) with AGI perl
?
I know how to get the caller id which is working fine
and is just
below:
code snip
Thanks in advance,
Angel.
use:
$exten = $input{'extension'};
to get the extension called.
Andy
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[Asterisk-Users] How to get the Called id with AGI
Hi all,
Is there a way to get the "called id" (the B number) with AGI perl ?
I know how to get the caller id which is working fine and is just below:
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$callerid = $input{'callerid'};
$AGI-say_digits($callerid);
}
Thanks in advance,
Angel.
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[Asterisk-Users] Run Asterisk without any .conf file ??
Hi all, I am very new with Astersik. Could some body tell me if it is possible to run Asterisk without any .conf file in /etc/asterisk ? I just want to test if my Asterisk has been installed correctly and as I am waiting for digium cards ... I have already tried but nothing happened after some verbose it stop... Thanks Angel __ Do you Yahoo!? Yahoo! Tax Center - File online by April 15th http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
