Re: [asterisk-users] sending sms from Asterisk server
Could you share your AGI script? CK On Wed, Aug 18, 2010 at 5:43 AM, Johann Hoehn johann.ho...@ecommerce.comwrote: On 08/17/2010 09:00 AM, Tino wrote: Hello, I would like to send sms to some external phone numbers from my asterisk server. Is it possible to send sms via softphones like X-Lite ? . Any tips regarding this will be helpful thanks This is easy to do by using email to SMS gateways. A list of them is on wikipedia (http://en.wikipedia.org/wiki/List_of_SMS_gateways). For the Asterisk side, you have an extension that sends the email. I personally use an AGI script for this part, but you could use a System() call as well. --johann -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending sms from Asterisk server
Johann Hoehn wrote: On 08/17/2010 09:00 AM, Tino wrote: Hello, I would like to send sms to some external phone numbers from my asterisk server. Is it possible to send sms via softphones like X-Lite ? . Any tips regarding this will be helpful thanks This is easy to do by using email to SMS gateways. A list of them is on wikipedia (http://en.wikipedia.org/wiki/List_of_SMS_gateways). For the Asterisk side, you have an extension that sends the email. I personally use an AGI script for this part, but you could use a System() call as well. Many telcos provide an SMSC, often also accessible over a landline. We use the Swisscom SMSC at 062210. (Swisscom subscription required). /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 debug of registration - Only getting RX and there is no TX response from Asterisk - is that normal?
Hi, Use requirecalltoken=no in your peer configuration Regards On Wed, Aug 11, 2010 at 4:28 AM, bruce bruce bruceb...@gmail.com wrote: Hello Everyone, I am trying to diagnose issue with my IAX2 extension not working. When I have iax2 set debug on all I see is this: *Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ * * Timestamp: 3ms SCall: 00130 DCall: 0 [64.229.229.111:64823]* * USERNAME: 100* * REFRESH : 60* * * *Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK* * Timestamp: 3ms SCall: 00130 DCall: 1 [64.229.229.111:64823]* So, all the packets are coming in, but there is no Tx response. Is that normal and is that how IAX2 works according to RFC to not respond back? I have checked my firewall and all is set fine. I have any WAN address to come in through port 4569 to map to the server and it worked last week but now it doesn't. Any suggestions? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL Connect problem...
Avoid to use MySQL dialplan application, instead write an AGI script for
this purpose
On Tue, Aug 17, 2010 at 4:59 PM, Geraint Lee gera...@gmail.com wrote:
Right, I'm baffled.
I have:
exten = s,1,MYSQL(Connect DB1 127.0.0.1 geraint xxx amis2)
exten = s,n,MYSQL(Query NORESULT ${DB1} INSERT\ INTO\ recordings\
(caller_number\,called_number\,date_created\,date_started\,in_use\,server_id)\
VALUES\ (\'${CALLERID(number)}\'\,\'${ARG1}\'\,NOW()\,NOW()\,\'Yes\'\,12))
exten = s,n,MYSQL(Query RESULT1 ${DB1} SELECT\ LAST_INSERT_ID())
exten = s,n,MYSQL(Fetch FOUND1 ${RESULT1} VALUE1)
exten = s,n,MYSQL(Clear ${RESULT1})
exten = s,n,MYSQL(Disconnect ${DB1})
exten = s,n,MixMonitor(${VALUE1}.wav)
exten = s,n,Set(CALLERID(all)=xxx)
exten = s,n,Dial(SIP/prov1/${ARG1})
in a macro to dial numbers...
Every few hours or so every call hangs on the s,1 MYSQL(Connect) and won't
work until i restart asterisk.
The mysql server has a maximum connections of 2048 (of which around 90 are
in use) so it's not a mysql connection limit problem from what i can tell
since while asterisk is stuck i can still log in to mysql just fine, as
can the web server.
Does anyone have any suggestions what could be causing asterisk to get
stuck here? i don't see anything in cli and core show channels just shows
everyone stuck in state ring on the connect string with no errors.
Cheers
Geraint
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[asterisk-users] Enhancing snmp mib
Hi, I'm quite pleased with the asterisk/res_snmp integration (at least a right one :) not some hackish scripted thingy) but i felt it's missing quite a few datas. What i would need is: * Per channels, number of inbound call received since asterisk startup (like a network interface) *number of outbound call sent * SIP status (peer reachable) * DAHDI show channels equivalent * queues status (number of calls proceeded, availability, ) * Custom oid, dialplan controlled, and/or dialpan readable * Accessing internal * database could be nice too For examples i would like to graph the numbers of calls received and sent on a channel type like a network interface (for monitoring PRI/BRI usage for exemple). However the current values is a realtime snapshot, not the best for this thing Is there already some work planned on the snmp interface ? regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Context
Unless I've got some massive misunderstanding, yes we do as described here http://www.voip-info.org/wiki/view/Asterisk+RealTime Why do you ask? On 17/08/10 17:27, Zeeshan Zakaria wrote: Ishfaq, do you use the asterisk real-time architecture? Zeeshan A Zakaria -- www.ilovetovoip.com http://www.ilovetovoip.com On 2010-08-17 11:45 AM, Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote: So that we can create a custom interface for our customers to be able to do simple stuff like adding extensions, changing dialplans without our (for our read my!) intervention. The DB storage is the main thing for us. On 17/08/10 16:09, Zeeshan Zakaria wrote: The whole point of using real-time, as the word 'rea... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Context
So why you need a reload when you are using the real-time architecture? Are you adding contexts in extensions.conf? If yes then where are you using the real-time? I asked because to me it seems you didn't understand what Dan is asking, and also seems you don't know how real-time works in regards to add contexts without reloading asterisk. On 2010-08-18 4:25 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Unless I've got some massive misunderstanding, yes we do as described here http://www.voip-info.org/wiki/view/Asterisk+RealTime Why do you ask? On 17/08/10 17:27, Zeeshan Zakaria wrote: Ishfaq, do you use the asterisk real-time architectu... -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Context
Hi I'll give you an example process flow Our customer logs onto our VoIP Portal and orders a SIP Geographic number Geo number is successfully ordered which inserts an entry into our extensions table in MySQL DB in the default context which has a goto command to a newly created specific context for handling the incoming on that number. The customer goes to the dial plan managing section of the portal and sets the number to ring an extension of his followed by going to voicemail. In the back end 2 new inserts are made into the extensions table for the newly created context. Also, an insert is made into a table to say that the extensions table has been updated. A cron runs and sees that the extensions table has been updated. this gets a distinct list of contexts from the extensions table and writes each one to a new extensions.con putting in the realtime switch for each context. The cron then executes a dialplan reload If you can see a more elegant way of doing the above so that no intervention is required by anyone other that the customer, please let me know as I'm always willing to learn better ways of doing things. Also, please bear in mind that we offer a hosted service for customers so we have lots of different companies all working off the same server(s) Ish On 18/08/10 10:37, Zeeshan Zakaria wrote: So why you need a reload when you are using the real-time architecture? Are you adding contexts in extensions.conf? If yes then where are you using the real-time? I asked because to me it seems you didn't understand what Dan is asking, and also seems you don't know how real-time works in regards to add contexts without reloading asterisk. On 2010-08-18 4:25 AM, Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote: Unless I've got some massive misunderstanding, yes we do as described here http://www.voip-info.org/wiki/view/Asterisk+RealTime Why do you ask? On 17/08/10 17:27, Zeeshan Zakaria wrote: Ishfaq, do you use the asterisk real-time architectu... -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor asterisk
man, see monast http://monast.sourceforge.net/ -- Renato dos Santos shazaum.wordpress.com 2010/8/17 Matt Riddell li...@venturevoip.com On 17/08/10 6:34 PM, Hans Witvliet wrote: On Mon, 2010-08-16 at 13:35 -0400, Jamie A. Stapleton wrote: Might be worth your time to check out: http://www.humbuglabs.org/ Though they write: ... insight into the enterprise’s telephony infrastructure. Utilizing a set of none-intrusive analytical technologies, Humbug is capable of interfacing directly with your PBX system, analyzing its traffic, plotting it and providing ... It looks (!) like an online-service. Who would give an outsider access to your phone-usage info? :) Take it you don't use Google Analytics, Facebook insights, Feedburner, Amazon EC3 etc etc. Sure you have to decide who you want to trust (personally I trust the humbuglabs guys) and what their level of protection is (are they looking after their own security), but it seems to be the way things are going at the mo. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Context
Thanks for the details. I Agreed you know what you are doing. As for doing it more elegantly, it is not possible to do a fruitful discussion without knowing all the details of how your dialplan works, neither is this the goal here. But I offer similar hosted PBX service, just use one context for all the companies, use various logics in the dialplan to differentiate between the tenants, their updates, new inserts etc., and never needed to do any reload. I identify tenant by their account ids and one context works for all. CDRs are perfectly fine, billing is even great and automated. If fact I have developed/programmed three multi-tenant solutions for far, including one for a busy client, and used the same logic. They all work perfectly fine. That was why I wondered why you needed reloads. But you must be doing something which genuinely requires a reload, not saying that you are not doing it right. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-18 6:49 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'll give you an example process flow Our customer logs onto our VoIP Portal and orders a SIP Geographic number Geo number is successfully ordered which inserts an entry into our extensions table in MySQL DB in the default context which has a goto command to a newly created specific context for handling the incoming on that number. The customer goes to the dial plan managing section of the portal and sets the number to ring an extension of his followed by going to voicemail. In the back end 2 new inserts are made into the extensions table for the newly created context. Also, an insert is made into a table to say that the extensions table has been updated. A cron runs and sees that the extensions table has been updated. this gets a distinct list of contexts from the extensions table and writes each one to a new extensions.con putting in the realtime switch for each context. The cron then executes a dialplan reload If you can see a more elegant way of doing the above so that no intervention is required by anyone other that the customer, please let me know as I'm always willing to learn better ways of doing things. Also, please bear in mind that we offer a hosted service for customers so we have lots of different companies all working off the same server(s) Ish On 18/08/10 10:37, Zeeshan Zakaria wrote: So why you need a reload when you are using the real... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor asterisk
Look into astassisstant. I don't remember the website, but google will take you their. It doesn't need any installations on the server, just a manager user in manager.conf. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-18 8:39 AM, Shazaum shaz...@gmail.com wrote: man, see monast http://monast.sourceforge.net/ -- Renato dos Santos shazaum.wordpress.com 2010/8/17 Matt Riddell li...@venturevoip.com On 17/08/10 6:34 PM, Hans Witvliet wrote: On Mon, 2010-08-16 at 13:35 -0400, Jamie A. Stapl... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending sms from Asterisk server
Hello Johann, Thanks for your advice in this matter. But i am not sure how to pass the numbers to be sent sms in the dialplan. On Wed, Aug 18, 2010 at 3:13 AM, Johann Hoehn johann.ho...@ecommerce.comwrote: On 08/17/2010 09:00 AM, Tino wrote: Hello, I would like to send sms to some external phone numbers from my asterisk server. Is it possible to send sms via softphones like X-Lite ? . Any tips regarding this will be helpful thanks This is easy to do by using email to SMS gateways. A list of them is on wikipedia (http://en.wikipedia.org/wiki/List_of_SMS_gateways). For the Asterisk side, you have an extension that sends the email. I personally use an AGI script for this part, but you could use a System() call as well. --johann -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 331 freezes connecting to FreePBX
On Tue, 2010-08-17 at 09:42 -1000, Ben Schorr wrote: Sorry, I should clarify - we have had a similar setup (IPSEC VPN, Polycom 331) working at a different location with a different handset for this same firm. We've never gotten the phone/VPN to work at this particular site. I was just trying to explain that it DOES appear to be successfully connecting to the TFTP server that provisions the phones. The TFTP server is sitting right next to the Asterisk server so if it can connect to one it should be able to connect to the other - basic connectivity, it appears, is working between the sites. We currently have 57 other Polycom phones (most of them 331s) working in this system, with the current application, just fine, including maybe 10 that connect over an IPSEC VPN from a 3rd location. That does give me an idea though...we could take the handset from the failing location to the 3rd location (barely a mile away) and plug it in and see if it works there. If it does then the problem must be somewhere in the connection and not with the handset itself. If it doesn't work in the other location either then the problem is probably with the phone and/or it's configuration. Make sense? Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Monday, August 16, 2010 11:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 331 freezes connecting to FreePBX On Mon, Aug 16, 2010 at 4:21 PM, Ben Schorr b...@rolandschorr.com wrote: We gave the phone a static IP address and pointed it to the configuration server on the remote end that has the CFG files for it. The phone starts up, downloads SIP and the new application and otherwise seems to be booting normally. Then it gets to the LAN Properties screen that shows the phone's IP address, MAC address and firmware version and then...nothing. It just sits there frozen. I have a suggestion... Put back the 'old application', and determine whether the 'new application' broke your phone boot. Since you don't mention changing anything else, survey says it's probably the last thing you changed that broke things. -- Ben, I have seen when trying to provision a Polycom phone over the WAN using tftp the phone lock up. Usually this is cause the TFTP transfer is crashing in the middle of one of the file transfers due the the nature (UDP). What I did to fix was use ftp or http over the WAN (TCP) or fire up a tftp server on the local LAN to do the initial provisioning of the phone, then point it back to the provisioning server across the WAN. Could be a problem with the link to. Never hurts to give it a try if you have another office so close. --Connor _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users attachment: winmail.dat-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Playing with sipvicious ..
... using it as a tool and understanding what it does... So one part of it's toolset identifys valid SIP accounts - and I was under the impression that alwaysauthreject=yes was supposed to stop this... However, it sends a request for a highly probably non-existent account, then sends requests for probably existing accounts and I guess compares the results - account not found vs. bad username or password... It thus trivially, and very quickly finds valid accounts when fed with a list of accounts to try in the first place (e.g. 100-999, or 1000-, etc.) I wonder if it's time to introduce yet another parameter to it - which will cause asterisk to return the same error code for all 3 conditions - and return the not found error, even on bad username or password. It breaks the RFC even more, but might it be worth it? (I've just had 30GB of sipvicious traffic sent to my hosted servers in a 12-hour period - it came from what looked like a VPS host in France - trivially firewalled out, but even dropping the packets didn't stop the flood! It's so badly written it appears to just ignore any return codes that it doesn't want, or even no returns at all!) Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: AsteriskNow REGISTER'ing s@ extension for all inbound trunks
Sending this to asterisk-users, in case anyone has AsteriskNOW experience they can share. Joe -- Forwarded message -- From: Joe Wood sch...@gmail.com Date: Wed, Aug 18, 2010 at 9:22 AM Subject: AsteriskNow REGISTER'ing s@ extension for all inbound trunks To: asterisk...@lists.digium.com Hello. Can someone tell me why AsteriskNow is reverting to registering s@ as an extension? 18 Aug 00:01:09.301/GLOBAL/ser: RECEIVED message from 209.221.186.51:5060: REGISTER sip:209.221.186.98 SIP/2.0 Via: SIP/2.0/UDP 209.221.186.51:5060;branch=z9hG4bK41fb6b8f;rport Max-Forwards: 70 From: sip:2063161...@209.221.186.98;tag=as7608 To: sip:2063161...@209.221.186.98 Call-ID: 5ada9ee829ddb4d311c5cb092b8d3...@209.221.186.50 CSeq: 104 REGISTER User-Agent: Asterisk PBX 1.6.2.11 Authorization: Digest username=2063161626, realm=pugetsoundtelecom.net, algorithm=MD5, uri=sip:209.221.186.98, nonce=4c6b8544f41a0643a65f4e17199268a018b32070, response=dcbb7adca43c6c7455c9942010c84423, qop=auth, cnonce=45cc3d6b, nc=0002 Expires: 120 Contact: sip:s...@209.221.186.51 Content-Length: 0 18 Aug 00:01:09.301/5ada9ee829ddb4d311c5cb092b8d3...@209.221.186.50/ser: processing REGISTER received from 209.221.186.51:5060 18 Aug 00:01:09.302/5ada9ee829ddb4d311c5cb092b8d3...@209.221.186.50/ser: saving contact sip:s...@209.221.186.51 into the database 18 Aug 00:01:09.302/GLOBAL/ser: SENDING message to 209.221.186.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 209.221.186.51:5060;branch=z9hG4bK41fb6b8f;rport=5060 From: sip:2063161...@209.221.186.98;tag=as7608 To: sip:2063161...@209.221.186.98;tag=5fceb36a80dbc27aca680924f1b8b505-19ad Call-ID: 5ada9ee829ddb4d311c5cb092b8d3...@209.221.186.50 CSeq: 104 REGISTER Contact: sip:s...@209.221.186.51;expires=300 Server: Sippy Softswitch v2.0.80 Content-Length: 0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: AsteriskNow REGISTER'ing s@ extension for allinbound trunks
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joe Wood Subject: [asterisk-users] Fwd: AsteriskNow REGISTER'ing s@ extension for allinbound trunks snip Since you can see the CLI log, please post your asterisk version (core show version) so we can see what flavor of Asterisk your AN is operating under. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending sms from Asterisk server
Le 18/08/2010 16:03, Tino a écrit : Hello Johann, Thanks for your advice in this matter. But i am not sure how to pass the numbers to be sent sms in the dialplan. agi(script,param1,param2,...,paramX) from your dialplan where script lies in /var/lib/asterisk/agi-bin On Wed, Aug 18, 2010 at 3:13 AM, Johann Hoehn johann.ho...@ecommerce.com mailto:johann.ho...@ecommerce.com wrote: On 08/17/2010 09:00 AM, Tino wrote: Hello, I would like to send sms to some external phone numbers from my asterisk server. Is it possible to send sms via softphones like X-Lite ? . Any tips regarding this will be helpful thanks This is easy to do by using email to SMS gateways. A list of them is on wikipedia (http://en.wikipedia.org/wiki/List_of_SMS_gateways). For the Asterisk side, you have an extension that sends the email. I personally use an AGI script for this part, but you could use a System() call as well. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WaitExten() always times out
Hi, My WaitExten() is not working as I expect it to. This is the relevant part of my context. It is meant to receive incoming calls. [incoming] exten = xxx,1,Background(hello-world) exten = xxx,2,WaitExten(7) exten = _X,1,AGI(myAGI.php) When I send the call from a .call, it works perfect, but when receiving an incoming call WaitExten() times out no matter what. [general] static=yes writeprotect=yes autofallthrough=yes clearglobalvars=no I experimented changing autofallthrough to no and got the same result. Any ideas about this strange behavior? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 debug of registration - Only getting RX and there is no TX response from Asterisk - is that normal?
That is set and here is what I get: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 01217 DCall: 0 [44.55.66.77:4569] USERNAME: 9988 REFRESH : 60 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 01217 DCall: 1 [44.55.66.77:4569] Any other suggestions. Anyone with a working pfsense configuration that can share with me? Thanks, Bruce On Wed, Aug 18, 2010 at 3:42 AM, Nasir Iqbal na...@ictinnovations.comwrote: Hi, Use requirecalltoken=no in your peer configuration Regards On Wed, Aug 11, 2010 at 4:28 AM, bruce bruce bruceb...@gmail.com wrote: Hello Everyone, I am trying to diagnose issue with my IAX2 extension not working. When I have iax2 set debug on all I see is this: *Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ * * Timestamp: 3ms SCall: 00130 DCall: 0 [64.229.229.111:64823] * * USERNAME: 100* * REFRESH : 60* * * *Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK* * Timestamp: 3ms SCall: 00130 DCall: 1 [64.229.229.111:64823] * So, all the packets are coming in, but there is no Tx response. Is that normal and is that how IAX2 works according to RFC to not respond back? I have checked my firewall and all is set fine. I have any WAN address to come in through port 4569 to map to the server and it worked last week but now it doesn't. Any suggestions? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending sms from Asterisk server
Un-top-posting...
On 08/17/2010 09:00 AM, Tino wrote:
I would like to send sms to some external phone numbers from my asterisk
server. Is it possible to send sms via softphones like X-Lite ? . Any
tips regarding this will be helpful
On Wed, Aug 18, 2010 at 3:13 AM, Johann Hoehn
johann.ho...@ecommerce.com wrote:
This is easy to do by using email to SMS gateways. A list of them is on
wikipedia (http://en.wikipedia.org/wiki/List_of_SMS_gateways). For the
Asterisk side, you have an extension that sends the email. I personally
use an AGI script for this part, but you could use a System() call as
well.
Using system() is almost always a hack -- and not the good kind :)
On Wed, 18 Aug 2010, Tino wrote:
Thanks for your advice in this matter. But i am not sure how to pass the
numbers to be sent sms in the dialplan.
You have a choice: you can pass them as channel variables or as command
line options. I use both, frequently in the same program. Unfortunately, I
can't clearly articulate why I use one over the other. If the variable is
something that exists for the life of the call like ${CLIENT-ID} I tend to
access it as a channel variable. If it's something that modifies the
behavior of the AGI (--debug or --verbose) I always pass it as a command
line option and use getopt_long()
First, you need to pick a language. If this is a SOHOish hobby project, it
doesn't matter -- pick a language you are comfortable with.
If this is a high volume, performance critical project -- I'd vote for c.
Once you've decided on a language, search out an established AGI library
and learn a bit about the protocol. It's very simple but not always
obvious. The 3 biggest stumbling blocks that trip up programmers are:
1) You have to read the AGI environment before anything else.
2) It's a request followed by a response. If you don't read the response,
bad things will happen.
3) It's STDIN/STDOUT based. If you try to debug by writing variables or
messages using echo/printf/puts/etc, bad things will happen.
Check out voip-info.org for more information on AGI.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
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[asterisk-users] WaitExten() always times out
Hi, My WaitExten() is not working as I expect it to. This is the relevant part of my context. It is meant to receive incoming calls. [incoming] exten = xxx,1,Background(hello-world) exten = xxx,2,WaitExten(7) exten = _X,1,AGI(myAGI.php) When I send the call from a .call, it works perfect, but when receiving an incoming call WaitExten() times out no matter what. [general] static=yes writeprotect=yes autofallthrough=yes clearglobalvars=no I experimented changing autofallthrough to no and got the same result. Any ideas about this strange behavior? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitExten() always times out
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones Subject: [asterisk-users] WaitExten() always times out Hi, My WaitExten() is not working as I expect it to. This is the relevant part of my context. It is meant to receive incoming calls. [incoming] exten = xxx,1,Background(hello-world) exten = xxx,2,WaitExten(7) exten = _X,1,AGI(myAGI.php) When I send the call from a .call, it works perfect, but when receiving an incoming call WaitExten() times out no matter what. snip I experimented changing autofallthrough to no and got the same result. Any ideas about this strange behavior? My best guess is that your problem is that _X isn't happy for whatever reason. Generally I use Waitexten for single digit processing like this Exten = 1234,1,goto(waitdtmf,s,1) [waitdtmf] Exten = s,1,background(hello-world) Exten = s,n,waitexten(7) Exten = 1,1,AGI(myAGI.php) Exten = 2,1,AGI(myAGI.php) Exten = I,1,playback(invalid) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: AsteriskNow REGISTER'ing s@ extension for allinbound trunks
sg01*CLI core show version Asterisk 1.6.2.11 built by root @ localhost.localdomain on a i686 running Linux on 2010-08-16 15:17:26 UTC On Wed, Aug 18, 2010 at 10:19 AM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joe Wood Subject: [asterisk-users] Fwd: AsteriskNow REGISTER'ing s@ extension for allinbound trunks snip Since you can see the CLI log, please post your asterisk version (core show version) so we can see what flavor of Asterisk your AN is operating under. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitExten() always times out
Thanks for you reply :). I thought of that and tried replacing _X with a numbers it should match (9), and it didn't work. It still times out as if no number was entered. On Wed, Aug 18, 2010 at 2:11 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Kathryn Jones *Subject:* [asterisk-users] WaitExten() always times out Hi, My WaitExten() is not working as I expect it to. This is the relevant part of my context. It is meant to receive incoming calls. [incoming] exten = xxx,1,Background(hello-world) exten = xxx,2,WaitExten(7) exten = _X,1,AGI(myAGI.php) When I send the call from a .call, it works perfect, but when receiving an incoming call WaitExten() times out no matter what. snip I experimented changing autofallthrough to no and got the same result. Any ideas about this strange behavior? My best guess is that your problem is that _X isn’t happy for whatever reason. Generally I use Waitexten for single digit processing like this Exten = 1234,1,goto(waitdtmf,s,1) [waitdtmf] Exten = s,1,background(hello-world) Exten = s,n,waitexten(7) Exten = 1,1,AGI(myAGI.php) Exten = 2,1,AGI(myAGI.php) Exten = I,1,playback(invalid) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitExten() always times out
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones Subject: Re: [asterisk-users] WaitExten() always times out Thanks for you reply :). I thought of that and tried replacing _X with a numbers it should match (9), and it didn't work. It still times out as if no number was entered. When you do the .call, it is probably on a local, SIP or IAX channel. When you hit the incoming, are you on a DAHDI channel? Also, a workaround would be to do Exten = t,1,AGI(myagi.php) So when the DTMF doesn't work it just drops through anyway. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR variables
Hello list!
I am trying to get hold of ${CDR(duration)} and ${CDR(billsec)} variables in
h
It seems that these variables always return 0. I am using Asterisk version
1.6.2.11. Can't I get these values other than using CDR reccords ??
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Re: [asterisk-users] WaitExten() always times out
My .call file goes out to a pstn number. That work around would be perfect :D, but I need the number given by the caller. On Wed, Aug 18, 2010 at 2:49 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Kathryn Jones *Subject:* Re: [asterisk-users] WaitExten() always times out Thanks for you reply :). I thought of that and tried replacing _X with a numbers it should match (9), and it didn't work. It still times out as if no number was entered. When you do the .call, it is probably on a local, SIP or IAX channel. When you hit the incoming, are you on a DAHDI channel? Also, a workaround would be to do Exten = t,1,AGI(myagi.php) So when the DTMF doesn’t work it just drops through anyway. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitExten() always times out
Hi, Are you sure asterisk is receiving and processing DMTF correctly? Are you using rfc2833, SIP INFO or inband DMTF? What is your asterisk version? I use WaitExten(5) all the time, no matter if they are single-digit or multiple-digit extensions. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 El 18/08/10 15:39, Kathryn Jones escribió: Thanks for you reply :). I thought of that and tried replacing _X with a numbers it should match (9), and it didn't work. It still times out as if no number was entered. On Wed, Aug 18, 2010 at 2:11 PM, Danny Nicholas da...@debsinc.com mailto:da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Kathryn Jones *Subject:* [asterisk-users] WaitExten() always times out Hi, My WaitExten() is not working as I expect it to. This is the relevant part of my context. It is meant to receive incoming calls. [incoming] exten = xxx,1,Background(hello-world) exten = xxx,2,WaitExten(7) exten = _X,1,AGI(myAGI.php) When I send the call from a .call, it works perfect, but when receiving an incoming call WaitExten() times out no matter what. snip I experimented changing autofallthrough to no and got the same result. Any ideas about this strange behavior? My best guess is that your problem is that _X isn’t happy for whatever reason. Generally I use Waitexten for single digit processing like this Exten = 1234,1,goto(waitdtmf,s,1) [waitdtmf] Exten = s,1,background(hello-world) Exten = s,n,waitexten(7) Exten = 1,1,AGI(myAGI.php) Exten = 2,1,AGI(myAGI.php) Exten = I,1,playback(invalid) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL Connect problem...
This is what I ended up doing, working fine now.
Cheers
On 18 August 2010 08:52, Nasir Iqbal na...@ictinnovations.com wrote:
Avoid to use MySQL dialplan application, instead write an AGI script for
this purpose
On Tue, Aug 17, 2010 at 4:59 PM, Geraint Lee gera...@gmail.com wrote:
Right, I'm baffled.
I have:
exten = s,1,MYSQL(Connect DB1 127.0.0.1 geraint xxx amis2)
exten = s,n,MYSQL(Query NORESULT ${DB1} INSERT\ INTO\ recordings\
(caller_number\,called_number\,date_created\,date_started\,in_use\,server_id)\
VALUES\ (\'${CALLERID(number)}\'\,\'${ARG1}\'\,NOW()\,NOW()\,\'Yes\'\,12))
exten = s,n,MYSQL(Query RESULT1 ${DB1} SELECT\ LAST_INSERT_ID())
exten = s,n,MYSQL(Fetch FOUND1 ${RESULT1} VALUE1)
exten = s,n,MYSQL(Clear ${RESULT1})
exten = s,n,MYSQL(Disconnect ${DB1})
exten = s,n,MixMonitor(${VALUE1}.wav)
exten = s,n,Set(CALLERID(all)=xxx)
exten = s,n,Dial(SIP/prov1/${ARG1})
in a macro to dial numbers...
Every few hours or so every call hangs on the s,1 MYSQL(Connect) and won't
work until i restart asterisk.
The mysql server has a maximum connections of 2048 (of which around 90 are
in use) so it's not a mysql connection limit problem from what i can tell
since while asterisk is stuck i can still log in to mysql just fine, as
can the web server.
Does anyone have any suggestions what could be causing asterisk to get
stuck here? i don't see anything in cli and core show channels just shows
everyone stuck in state ring on the connect string with no errors.
Cheers
Geraint
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Re: [asterisk-users] WaitExten() always times out
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones Subject: Re: [asterisk-users] WaitExten() always times out My .call file goes out to a pstn number. That work around would be perfect :D, but I need the number given by the caller. My bet is that the pstn/DAHDI delay is eating part of your message (it takes 3-7 seconds from Dial to actually connect). Try putting a wait(5) in front of the Background command. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR variables
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada
Subject: [asterisk-users] CDR variables
Hello list!
I am trying to get hold of ${CDR(duration)} and ${CDR(billsec)} variables
in h
It seems that these variables always return 0. I am using Asterisk version
1.6.2.11. Can't I get these values other than using CDR reccords ??
In cdr.conf, is endbeforehexten=yes ?
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Re: [asterisk-users] WaitExten() always times out
I must not be receiving them properly, since I can't make it work. I just can't see why :P. My asterisk version is 1.6.2.6. Like I said before, for outgoing .call files WaitExten works fine, it's on incoming calls that I cannot receive the number I need. I had not checked my dtmf mode, this is new to me. So I was using asterisk default rfc2833. I am making pstn calls from regular telephones, through asterisk. What dtmfmode should I use? Could that be my problem? On Wed, Aug 18, 2010 at 2:57 PM, Miguel Molina mmol...@millenium.com.cowrote: Hi, Are you sure asterisk is receiving and processing DMTF correctly? Are you using rfc2833, SIP INFO or inband DMTF? What is your asterisk version? I use WaitExten(5) all the time, no matter if they are single-digit or multiple-digit extensions. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 El 18/08/10 15:39, Kathryn Jones escribió: Thanks for you reply :). I thought of that and tried replacing _X with a numbers it should match (9), and it didn't work. It still times out as if no number was entered. On Wed, Aug 18, 2010 at 2:11 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Kathryn Jones *Subject:* [asterisk-users] WaitExten() always times out Hi, My WaitExten() is not working as I expect it to. This is the relevant part of my context. It is meant to receive incoming calls. [incoming] exten = xxx,1,Background(hello-world) exten = xxx,2,WaitExten(7) exten = _X,1,AGI(myAGI.php) When I send the call from a .call, it works perfect, but when receiving an incoming call WaitExten() times out no matter what. snip I experimented changing autofallthrough to no and got the same result. Any ideas about this strange behavior? My best guess is that your problem is that _X isn’t happy for whatever reason. Generally I use Waitexten for single digit processing like this Exten = 1234,1,goto(waitdtmf,s,1) [waitdtmf] Exten = s,1,background(hello-world) Exten = s,n,waitexten(7) Exten = 1,1,AGI(myAGI.php) Exten = 2,1,AGI(myAGI.php) Exten = I,1,playback(invalid) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Context
We don't use a context for that. We set up dialplan code in a non asterisk part of MySQL called routing types. When a customer selects a DDI number they can choose a routing type to use with it. These routing types allow for variable substitution - i.e. if someone adds the routing type Direct Routing With Failover, there is a variable with this type called failover routing. The call is then sent to their SIP or IAX2 device (if not on the same machine that has the DDI number it uses DUNDI to find the appropriate machine). If somebody is unavailable on all machines, it sets the account code to the customer and goes to the outbound context for dialling. The difference is that in the routing type we don't use extensions (just applications). When someone adds the routing type to their DDI realtime extensions are created with extension 5551234 (or whatever their DDI number is) and priorities which increase. It makes some things a bit harder, but you can always use labels and the read application. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Context
Hi Matt, That's somewhat closer to what I do in my dialplan as well. But Dan apparantly wants to add new contexts in his `extensions` table. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-18 6:02 PM, Matt Riddell li...@venturevoip.com wrote: We don't use a context for that. We set up dialplan code in a non asterisk part of MySQL called routing types. When a customer selects a DDI number they can choose a routing type to use with it. These routing types allow for variable substitution - i.e. if someone adds the routing type Direct Routing With Failover, there is a variable with this type called failover routing. The call is then sent to their SIP or IAX2 device (if not on the same machine that has the DDI number it uses DUNDI to find the appropriate machine). If somebody is unavailable on all machines, it sets the account code to the customer and goes to the outbound context for dialling. The difference is that in the routing type we don't use extensions (just applications). When someone adds the routing type to their DDI realtime extensions are created with extension 5551234 (or whatever their DDI number is) and priorities which increase. It makes some things a bit harder, but you can always use labels and the read application. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Pr... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IXJ issues on 1.4.35
My thanks for previous help on fixing IXJ issues in 1.2.40; I now have problems with a just-built 1.4.35 on the same host: [Aug 18 17:26:48] WARNING[27209]: app_dial.c:1298 dial_exec_full: Unable to create channel of type 'Phone' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) O/S: Linux 2.4.27; IXJ driver for Linux Rev. 3.5, gcc 3.01 I applied the patch for dialtone as per my issue on 1.2.40. Build log contains no warnings for chan_phone. Build tested using install to DESTDIR=/tmp/asterisk; make samples; edited 'extensions.conf' to uncomment extension '1265' (Phone/phone0); added skinny extension to test Cisco 7920 (which worked except for audio stalling): exten = 1266,1,Dial(Skinny/1...@7920-1) exten = 1266,n,Goto(s,5) Asterisk started with -vvv -C /tmp/asterisk/etc/asterisk/asterisk.conf and controlled from 'asterisk -r' What other debugging should I enable? Again, all help is much appreciated. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IXJ issues on 1.4.35
On Wed, 18 Aug 2010, Infra wrote: My thanks for previous help on fixing IXJ issues in 1.2.40; I now have problems with a just-built 1.4.35 on the same host: [Aug 18 17:26:48] WARNING[27209]: app_dial.c:1298 dial_exec_full: Unable to create channel of type 'Phone' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) O/S: Linux 2.4.27; IXJ driver for Linux Rev. 3.5, gcc 3.01 I applied the patch for dialtone as per my issue on 1.2.40. Build log contains no warnings for chan_phone. Build tested using install to DESTDIR=/tmp/asterisk; make samples; edited 'extensions.conf' to uncomment extension '1265' (Phone/phone0); added skinny extension to test Cisco 7920 (which worked except for audio stalling): exten = 1266,1,Dial(Skinny/1...@7920-1) exten = 1266,n,Goto(s,5) Asterisk started with -vvv -C /tmp/asterisk/etc/asterisk/asterisk.conf and controlled from 'asterisk -r' What other debugging should I enable? Again, all help is much appreciated. Jumpped the gun -- I didn't finish editing the sample 'phone.conf' file; the fxs line is now working, but without dialtone. I will revert to the unpatched version of chan_phone.c and test again. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IXJ issues on 1.4.35
On Wed, 18 Aug 2010, Infra wrote: On Wed, 18 Aug 2010, Infra wrote: My thanks for previous help on fixing IXJ issues in 1.2.40; I now have problems with a just-built 1.4.35 on the same host: snip I applied the patch for dialtone as per my issue on 1.2.40. snip the fxs line is now working, but without dialtone. I will revert to the unpatched version of chan_phone.c and test again. No dialtone with the unpatched version. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sip peers : musiconhold class
Hi to convert wav file use following sox 'orgFile' -w -r 8000 -c 1 -s 'fixedFile' while replace orgFile and fixedFile with actual filenames If still now luck try with mp3 Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending sms from Asterisk server
I would rather use .call files. So easy to produce a text file...
On 18 August 2010 21:02, Steve Edwards asterisk@sedwards.com wrote:
Un-top-posting...
On 08/17/2010 09:00 AM, Tino wrote:
I would like to send sms to some external phone numbers from my asterisk
server. Is it possible to send sms via softphones like X-Lite ? . Any tips
regarding this will be helpful
On Wed, Aug 18, 2010 at 3:13 AM, Johann Hoehn johann.ho...@ecommerce.com
wrote:
This is easy to do by using email to SMS gateways. A list of them is on
wikipedia (http://en.wikipedia.org/wiki/List_of_SMS_gateways). For the
Asterisk side, you have an extension that sends the email. I personally use
an AGI script for this part, but you could use a System() call as well.
Using system() is almost always a hack -- and not the good kind :)
On Wed, 18 Aug 2010, Tino wrote:
Thanks for your advice in this matter. But i am not sure how to pass the
numbers to be sent sms in the dialplan.
You have a choice: you can pass them as channel variables or as command
line options. I use both, frequently in the same program. Unfortunately, I
can't clearly articulate why I use one over the other. If the variable is
something that exists for the life of the call like ${CLIENT-ID} I tend to
access it as a channel variable. If it's something that modifies the
behavior of the AGI (--debug or --verbose) I always pass it as a command
line option and use getopt_long()
First, you need to pick a language. If this is a SOHOish hobby project, it
doesn't matter -- pick a language you are comfortable with.
If this is a high volume, performance critical project -- I'd vote for c.
Once you've decided on a language, search out an established AGI library
and learn a bit about the protocol. It's very simple but not always obvious.
The 3 biggest stumbling blocks that trip up programmers are:
1) You have to read the AGI environment before anything else.
2) It's a request followed by a response. If you don't read the response,
bad things will happen.
3) It's STDIN/STDOUT based. If you try to debug by writing variables or
messages using echo/printf/puts/etc, bad things will happen.
Check out voip-info.org for more information on AGI.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] channel variables in AGI
On Wednesday, August 11, 2010 11:08:37 am Tino wrote: #!/bin/bash -x T=$agi_uniqueid I want to save value of 'agi_uniqueid' channel variable into a variable called 'T' in my script When executing and AGI from the dialplan, it will dump out it's variables immediately, so you need to tell Bash to read them in and write them to whatever variables you want. For example, see: http://messinet.com/trac/asterisk-fax-gw/browser/fax-gw.agi#L622 Here, I set the variable name from Asterisk to the variable value from Asterisk. So I end up with: agi_uniqueid=123456... (or whatever the uniqueid was) Then I could go on to say T=$agi_uniqueid -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending sms from Asterisk server
On 08/19/2010 08:21 AM, Tiago Geada wrote:
I would rather use .call files. So easy to produce a text file...
On 18 August 2010 21:02, Steve Edwards asterisk.org
http://asterisk.org@sedwards.com http://sedwards.com wrote:
Un-top-posting...
On 08/17/2010 09:00 AM, Tino wrote:
I would like to send sms to some external phone numbers from
my asterisk server. Is it possible to send sms via softphones
like X-Lite ? . Any tips regarding this will be helpful
On Wed, Aug 18, 2010 at 3:13 AM, Johann Hoehn
johann.ho...@ecommerce.com
mailto:johann.ho...@ecommerce.com wrote:
This is easy to do by using email to SMS gateways. A list of
them is on wikipedia
(http://en.wikipedia.org/wiki/List_of_SMS_gateways). For the
Asterisk side, you have an extension that sends the email. I
personally use an AGI script for this part, but you could use
a System() call as well.
Using system() is almost always a hack -- and not the good kind :)
On Wed, 18 Aug 2010, Tino wrote:
Thanks for your advice in this matter. But i am not sure how
to pass the numbers to be sent sms in the dialplan.
You have a choice: you can pass them as channel variables or as
command line options. I use both, frequently in the same program.
Unfortunately, I can't clearly articulate why I use one over the
other. If the variable is something that exists for the life of
the call like ${CLIENT-ID} I tend to access it as a channel
variable. If it's something that modifies the behavior of the AGI
(--debug or --verbose) I always pass it as a command line option
and use getopt_long()
First, you need to pick a language. If this is a SOHOish hobby
project, it doesn't matter -- pick a language you are comfortable
with.
If this is a high volume, performance critical project -- I'd vote
for c.
Once you've decided on a language, search out an established AGI
library and learn a bit about the protocol. It's very simple but
not always obvious. The 3 biggest stumbling blocks that trip up
programmers are:
1) You have to read the AGI environment before anything else.
2) It's a request followed by a response. If you don't read the
response, bad things will happen.
3) It's STDIN/STDOUT based. If you try to debug by writing
variables or messages using echo/printf/puts/etc, bad things will
happen.
Check out voip-info.org http://voip-info.org for more
information on AGI.
Hi,
how do you get the text to send?
text that is sent from X-Lite for example.
thx,
anton
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