Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work
2013/8/1 Joshua Colp jc...@digium.com Larry Moore wrote: On 31/07/2013 8:08 PM, Joshua Colp wrote: Zoltán Fekete wrote: Thank You Larry! I have discussed with my provider. They are not able to insert the T38MaxBitRate value into the sip answer. :( https://gist.github.com/**anonymous/6120148https://gist.github.com/anonymous/6120148(line 559) That means we are not able to passtrough T38 Faxes with any asterisk version at all? What do you mean? Am I able to modify and compile the source? Is it compicated? (I'm not a developer) Based on the SDP in your gist the remote implementation has given no attributes with the T.38 stream which makes it pretty broken (T38FaxRateManagement is mandatory) and fun. The two hard parts really would be 1. Modifying Asterisk in a sane fashion to cope and 2. Determining the exact settings to make the implementation happy. Defaults as defined in the spec are fine and good, but my experience has taught me to throw those out the window when it comes to actual implementations. It would seem that having a configurable option would be an idea for this scenario. That implies it would solve the problem, which my gut and experience tells me... it wouldn't. I think the T.38 implementation is just cobbled together and without knowing exactly how it behaves getting it to work would likely be a nightmare (trust me, I've spent time in those deep dark reaches). Throwing assumptions and defaults at it to try to make it work is of course an option. My testing with Asterisk 1.8 and T.38, I obserevd that setting FAXOPT(minrate) or FAXOPT(maxrate)had no effect, I concluded that when Astrerisk is receiving it uses hard coded values - is this a sane thing to do?! When Asterisk is receiving the stack implementation offers what it wants, with the ability to override. So Asterisk doesn't hard code those values, the stack provides them. What is hard coded is the default values if none are received. I would even say it's a bug that the negotiation doesn't fail, since the remote side isn't providing a mandatory attribute. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- __**__**_ Yes you're right! As I know FAXOPT() value affect only when asterisk woks as gateway. We need passtrouh because my endpoints and also my provider supports T38. https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway Using T.38 Gateway mode T.38 Gateway mode should be used when one leg of a call is not capable of T.38 mode. In the event that both legs are capable and Gateway mode is configured, then the Gateway will step out of the way, allowing *transparent T.38 passthrough*. The main problem is that I can't use G711 for the entire fax session because the endpoints has 20-30ms response time. When I try to use my Asterisk as FAXOPT gateway (endpoint leg T38 and provider leg G711) can I force somehow to not accept the T38 re-INVITEs from the provider? They have ~1ms response time, so G711 on that leg would be fine but they also detect fax CED tones and sends the re-INVITEs. Regards, Blaxy -- _ -- 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] Asterisk - ODBC engine not available
On Wed, Jul 31, 2013 at 2:19 AM, Prashant A. abhang_prash...@yahoo.co.inwrote: But it still show me the following error [Jul 31 12:36:18] WARNING[1381]: config.c:2256 find_engine: Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not available Can someone help me on this Grab a log showing VERBOSE,DEBUG,WARNING,NOTICE https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information If you grab it from the CLI be sure VERBOSE and DEBUG are turned up to 5. Pastebin that and we'll be able to see a little better what is going on hopefully. Give us the output of core show version as well. -- Rusty Newton Digium, Inc. | Community Support Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USdirect: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Question Asterisk Manager
Hi A small question on Asterisk Manager. I use Perl Script for start a call: my $response = $astman-sendcommand( Action = 'Originate', Channel = 'SIP/ASTERISK/$Extension', Exten = '200', Context = 'MyContext', Priority = '1', Async = '1' ); That's start the call, but only the position of the corresponding sounds departing. As soon as he clinched, that the second ringing phone. Is there a way for two phone ring at the same time? Thanks Olivier -- _ -- 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] asterisk and IVR
i have Create a h extension and all works without issue .thank you so
much for your help and support i really appreciate it.
2013/7/31 A J Stiles asterisk_l...@earthshod.co.uk
On Wednesday 31 July 2013, Salaheddine Elharit wrote:
hi
i use the code below but i didn't get the We reached step 102 the same
result
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,GotoIf($[${DIALSTATUS} = ANSWER]?answered)
exten = 534,n,Goto(home,s,1)
exten = 534,n(answered),NoOp(Call was answered)
exten = 534,102,NoOp(We reached step 102)
So it looks as though it's breaking out of the extension logic altogether,
if
the call gets answered. In that case, you'll have to do it the
old-fashioned
way: Create a h extension (which fires when a call is hung up) *in the
same context as your 534 extension* (you can have a h extension in each
context, if needs be), and do all your fancy end-of-call stuff there.
exten = 534,1,Dial(SIP/228, 10)
exten = 534,n,NoOp(Dial status is ${DIALSTATUS})
exten = 534,n,Goto(home,s,1)
exten = h,1,NoOp(Hangup received. Dial status is ${DIALSTATUS})
Note that if there are other extensions in the context, h will be called
when
they get hung up -- you might need some logic in there to deal with this
(or
cheat by just having one extension besides h in this context, and use a
fully-
specified Goto() to jump into it.)
--
AJS
Answers come *after* questions.
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[asterisk-users] Asterisk 1.4 CDR vs VoIP Innovations CDR
When I compare my total minutes on the bill from VoIP Innovations, to the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count of minutes. I'm wondering why it's there. Are there different methods of counting the billable start or end point of a phone call? If it matters, I'm counting more termination minutes than they are and they're counting more origination minutes than I am. -- _ -- 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] asterisk and IVR
On Thursday 01 August 2013, Salaheddine Elharit wrote: i have Create a h extension and all works without issue .thank you so much for your help and support i really appreciate it. Good -- glad you got it working. But in future, please remember to type your reply *after* the thing you are replying to, not before it. That way, the logical flow of the conversation can be maintained in the correct order, with answers appearing after questions. -- AJS Answers come *after* questions. -- _ -- 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] Need to figure out DAHDI logical group from CDR record
I have a bunch of CDR records in the mysql database asteriskcdrdb on a FreePBX system. There is a DAHDI trunk defined in FreePBX which uses the gN identifier to make calls. So in this setup the trunk is roughly equivalent to a DAHDI logical group. I want to know, given a CDR, which logical group (and therefore, trunk) was used to place the call, or was used to receive the call. In Asterisk 1.6 and lower, the DAHDI channel encoded the B channel, as in DAHDI/aaa-bbb where aaa was the ISDN B channel. Since, as far as I can see, the logical group groups channels together, the strategy was to ask Asterisk for the dahdi groups, list the B channels, and fetch the records that referenced those B channels. In Asterisk 1.8, this no longer works. Since the DAHDI channel name no longer references a B channel ( https://issues.asterisk.org/jira/browse/ASTERISK-17229) I can no longer extract the B channel to associate a particular CDR with a particular logical group, Though, I am not particularly interested in the B channel, except as a means to build the association. Is there a way to guess the logical group given the standard CDR fields? All of the strategies I can think of require adding new fields to asteriskcdrdb, which does nothing to help me with the CDR records already written. The strategies for future association involve using the dialplan to capture CHANNEL(dahdi_span/dahdi_channel/dahdi_type) and writing this information in additional fields in asteriskcdrdb. However, some field to inform the logical group would be more useful for me. -- _ -- 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] Asterisk 1.4 CDR vs VoIP Innovations CDR
May be as simple as this: When you terminate a call you start the call before they even get it. When they originate a call, they start the call before you get it. Just a guess without really thinking about this too much. On Thu, Aug 1, 2013 at 10:28 AM, Adam Moffett adamli...@plexicomm.netwrote: When I compare my total minutes on the bill from VoIP Innovations, to the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count of minutes. I'm wondering why it's there. Are there different methods of counting the billable start or end point of a phone call? If it matters, I'm counting more termination minutes than they are and they're counting more origination minutes than I am. -- __**__**_ -- 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-**usershttp://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
[asterisk-users] Local agent/member in-use after transfer
I currently have all agents/members logged in with local channels. When a call is sent to one of the agents, then the agent transfers the call out the line frees up on their phone but still shows in-use until the call that was transferred is hung up. How can I free up the agent/local channel when the call is transferred? This is a huge problem because the agent can no longer receive calls on their extension. If they are the only agent logged in, then no other calls can be answered. If the transferred calls last an hour then no calls can be answered by this agent for an hour. I know I can set ringinuse=yes but this causes the agent to be interrupted while on calls which is not the desired result. -- _ -- 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] Asterisk 1.4 CDR vs VoIP Innovations CDR
A fair guess May be as simple as this: When you terminate a call you start the call before they even get it. When they originate a call, they start the call before you get it. Just a guess without really thinking about this too much. On Thu, Aug 1, 2013 at 10:28 AM, Adam Moffett adamli...@plexicomm.net mailto:adamli...@plexicomm.net wrote: When I compare my total minutes on the bill from VoIP Innovations, to the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count of minutes. I'm wondering why it's there. Are there different methods of counting the billable start or end point of a phone call? If it matters, I'm counting more termination minutes than they are and they're counting more origination minutes than I am. -- _ -- 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
[asterisk-users] External sip phones register with the servers IP...
We have just updated our office server to Asterisk 11.4.0 from 1.8.15 and internally everything is working fine. The problem we are having is that we cannot use any external phone connected through the Internet. This used to work fine with 1.8 but since the upgrade whenever you register any phone from an outside network the phone tries to register using the servers internal IP. I endo up having something like this: Sending to 187.163.93.235:58545 (no NAT) -- Registered SIP '2003' at 192.168.2.50:58545 Reliably Transmitting (no NAT) to 192.168.2.50:58545: OPTIONS sip:2003@192.168.2.50:58545;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.2.50:5060;branch=z9hG4bK5f2019c0 Max-Forwards: 70 From: asterisk sip:asterisk@192.168.2.50;tag=as4ed13172 To: sip:2003@192.168.2.50:58545;ob Contact: sip:asterisk@192.168.2.50:5060 Call-ID: 46fd0ef840d6781d219269ae415e156e@192.168.2.50:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.4.0 Date: Fri, 02 Aug 2013 00:27:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 I really cannot understand what is wrong, I have checked my sip.conf configuration and it is the same as in past versions. externaddr and localnet are set to the proper values. Any ideas? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -- _ -- 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] External sip phones register with the servers IP...
- Original Message - From: Carlos Chavez cur...@telecomabmex.com To: asterisk-users@lists.digium.com Sent: Thursday, August 1, 2013 8:41:19 PM Subject: [asterisk-users] External sip phones register with the servers IP... We have just updated our office server to Asterisk 11.4.0 from 1.8.15 and internally everything is working fine. The problem we are having is that we cannot use any external phone connected through the Internet. This used to work fine with 1.8 but since the upgrade whenever you register any phone from an outside network the phone tries to register using the servers internal IP. I endo up having something like this: Sending to 187.163.93.235:58545 (no NAT) -- Registered SIP '2003' at 192.168.2.50:58545 Reliably Transmitting (no NAT) to 192.168.2.50:58545: OPTIONS sip:2003@192.168.2.50:58545;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.2.50:5060;branch=z9hG4bK5f2019c0 Max-Forwards: 70 From: asterisk sip:asterisk@192.168.2.50;tag=as4ed13172 To: sip:2003@192.168.2.50:58545;ob Contact: sip:asterisk@192.168.2.50:5060 Call-ID: 46fd0ef840d6781d219269ae415e156e@192.168.2.50:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.4.0 Date: Fri, 02 Aug 2013 00:27:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 I really cannot understand what is wrong, I have checked my sip.conf configuration and it is the same as in past versions. externaddr and localnet are set to the proper values. Any ideas? Did you look at the CHANGES file? There are new settings for NAT. If you are using the same settings as in 1.8, there is a posiblity that you will have problems depending on what settings you have (which you did not include in this message). Also, I would recommend 11.5 since there was a one-way audio issue fixed related to using the two new NAT settings. -- Michael (elguero) -- _ -- 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] Asterisk 1.4 CDR vs VoIP Innovations CDR
On Thu, Aug 1, 2013 at 10:28 AM, Adam Moffett adamli...@plexicomm.netwrote: When I compare my total minutes on the bill from VoIP Innovations, to the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count of minutes. I'm wondering why it's there. Are there different methods of counting the billable start or end point of a phone call? If it matters, I'm counting more termination minutes than they are and they're counting more origination minutes than I am. If I remember correctly, they bill in sub-minute increments, something like 60 second minimum, then every 6 seconds after that. In other words, if you have a 20 second call, it's billed as 60 seconds, however, if you have a 62 second call, it's billed as 66. I don't remember what they're specific increments are, but I don't believe it was a straight bill. Are you finding that you're off by just a few seconds per call, or by minutes? When you say you're off by 3-4%, are you saying your CDR reports 100 minutes on a call and they are showing 104 minutes, or vice versa? -- _ -- 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] Local agent/member in-use after transfer
From: tjrl...@live.com To: asterisk-users@lists.digium.com Date: Thu, 1 Aug 2013 12:50:32 -0500 Subject: [asterisk-users] Local agent/member in-use after transfer I currently have all agents/members logged in with local channels. When a call is sent to one of the agents, then the agent transfers the call out the line frees up on their phone but still shows in-use until the call that was transferred is hung up. How can I free up the agent/local channel when the call is transferred? This is a huge problem because the agent can no longer receive calls on their extension. If they are the only agent logged in, then no other calls can be answered. If the transferred calls last an hour then no calls can be answered by this agent for an hour. I know I can set ringinuse=yes but this causes the agent to be interrupted while on calls which is not the desired result. -- _ -- 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 I found a solution and wanted to post it for those that may run into this trouble in the future. I use the manager interface to login my agents using a web page. After much digging I finally found the StateInterface: option available in 1.6 and above. I added it to my PHP login screen like this.. fputs($socket2, StateInterface: SIP/.$agentid.\r\n); The problem is that the queue was monitoring the local channel in terms of when a call was hungup or not, allowing other calls to come through. When a transfer happened the Local channel was not released. Adding the StateInterface option apparently allows the queue to monitor the actual channel, not the local channel. I couldn't find much documentation on this option, just stumbled upon it. Fixed my issue though! Thought I would add to the little info that seems to be out there about this option. -- _ -- 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
