Re: [asterisk-users] zaptel/dahdi settings for singtel E1 line
For India following are the parameters that i used to configure with sangoma E1 cards, coding - HDB3 framing - CRC4 or Non-CRC depends upon the service provider line type - national switchtype - EuroISDN On Thu, Feb 10, 2011 at 11:20 AM, Faisal Hanif fai...@vopium.com wrote: The settings you are asking varies in different countries and providers. You need to contact you provider for it. *From:* Roi Stork roi.st...@gmail.com *Sent:* Thursday, February 10, 2011 9:49 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Subject:* [asterisk-users] zaptel/dahdi settings for singtel E1 line Anyone here who has configured zaptel/dahdi for a singtel E1 line? What are the settings for coding, framing, line type and switchtype? -- -- _ -- 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 -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:sai...@gtalk2voip.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] Unable to make outgoing calls with Internode
On Thu, 10 Feb 2011 13:08:29 +1000, Da Rock asterisk-us...@herveybayaustralia.com.au wrote: I have an asterisk 1.8 server running on FreeBSD 8.1, and another FreeBSD 8.1 running as a firewall/gateway with PF. Does it work if you remove the firewall from the equation? Since Internode is an OZ company, and provided this issue turns out to be specific to that provider, you might have more luck solving the problem by asking in the Whirlpool forum: http://forums.whirlpool.net.au/forum/68 -- _ -- 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] Question about EuroBRI final 2 digits
This sounds like a job for DISA. http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA to see if DISA helps. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: 03 February 2011 19:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question about EuroBRI final 2 digits Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call before they finish dialing all 8 digits. Is there a way to prevent this, or to catch the additional 2 digits somewhere in the stream? The receptionist is unhappy because she gets all the 6-digit calls and must then transfer. Is this a p2p vs p2mp issue? Thanks in advance, Cassius Smith If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Early audio SIP sequence order question
Hello We have quite some problems with early audio with our asterisk 1.6.2.15 What we observe is: Asterisk - Carrier PBX Asterisk:Invite(+sdp) = Carrier Carrier starts to send RTP Audio (ignored by Asterisk) Asterisk = Carrier:100 Trying Asterisk = Carrier:180 Ringing Asterisk signals Ringing to the caller which in turn generated the ringing tone (still ignoring the early audio sent by the carrier). Asterisk = Carrier:200 OK(+sdp) Asterisk:ACK = Carrier Asterisk starts to send RTP Audio to Carrier Only now Asterisk starts playing Audio to the caller. This causes quite troubles, as the price of a value added number is announced in early audio in switzerland, giving the caller a chance to hang up before the call is established. But the caller connected to asterisk does not hear that early audio announcement. Is this an asterisk bug, or should the carrier have signaled 183 Session Progress instead of 180 Ringing? Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz Web http://www.imp.ch __ -- _ -- 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] Early audio SIP sequence order question
Go here http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial and use proper parameters to dial command to pass early media. -Original Message- From: Benoit Panizzon Sent: Thursday, February 10, 2011 4:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Early audio SIP sequence order question Hello We have quite some problems with early audio with our asterisk 1.6.2.15 What we observe is: Asterisk - Carrier PBX Asterisk:Invite(+sdp) = Carrier Carrier starts to send RTP Audio (ignored by Asterisk) Asterisk = Carrier:100 Trying Asterisk = Carrier:180 Ringing Asterisk signals Ringing to the caller which in turn generated the ringing tone (still ignoring the early audio sent by the carrier). Asterisk = Carrier:200 OK(+sdp) Asterisk:ACK = Carrier Asterisk starts to send RTP Audio to Carrier Only now Asterisk starts playing Audio to the caller. This causes quite troubles, as the price of a value added number is announced in early audio in switzerland, giving the caller a chance to hang up before the call is established. But the caller connected to asterisk does not hear that early audio announcement. Is this an asterisk bug, or should the carrier have signaled 183 Session Progress instead of 180 Ringing? Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz Web http://www.imp.ch __ -- _ -- 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] Question about EuroBRI final 2 digits
Hello, Maybe try that: In your incoming isdn context: [isdn-incoming] exten = s,1,Set(TIMEOUT(digits)=3) exten = s,2,WaitExten(2) exten = s,3,Dial(SIP/operator...) exten = 10,1,Dial(SIP/10) exten = 20,1,Dial(SIP/20) So if a call comes in Asterisk waits, 2 seconds for further digits dialed and if so jumps to the right extension in the context. Overlapdial should be yes. yours christian gansberger www.accm.at On 3 February 2011 20:45, Cassius Smith cass...@cassius.org wrote: Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call before they finish dialing all 8 digits. Is there a way to prevent this, or to catch the additional 2 digits somewhere in the stream? The receptionist is unhappy because she gets all the 6-digit calls and must then transfer. Is this a p2p vs p2mp issue? Thanks in advance, Cassius Smith -- _ -- 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] CDR with unix time.
Good morning everyone. I wonder if it is possible, without touching the source code, to Asterisk save the cdr with date in unix time instead of the default date. It's possible? Thanks in advance, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.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] Early audio SIP sequence order question
Hi Faisal http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Thank you, but I seem to miss the option which tells asterisk to pass audio even if no 183 or 200 is received. No, we don't set the 'r' Flag while dialing out. So, my question ist sill the same. Sould asterisk pass audio of it didn't yet receive a 183 or 200 message, or is the carrier doing wrong in sending early audio without 183? Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz Web http://www.imp.ch __ -- _ -- 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 with unix time.
Well. I suggest to use DB function instead of modifying asterisk source. You can add one additional column and write and after-insert trigger in your cdrs table which convert dattime to your required format and update the value of added column. From: Rodrigo Lang Sent: Thursday, February 10, 2011 5:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CDR with unix time. Good morning everyone. I wonder if it is possible, without touching the source code, to Asterisk save the cdr with date in unix time instead of the default date. It's possible? Thanks in advance, -- Rodrigo Lang Opening your mind - Just another Open Source site -- _ -- 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] CDR with unix time.
Just use uniqueid, which is exactly what you want. No modification is necessary. Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com Find us on Facebook http://www.facebook.com/pages/Vilnius-Lithuania/Kolmisoft/106746839379147 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rodrigo Lang Sent: Thursday, February 10, 2011 2:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CDR with unix time. Good morning everyone. I wonder if it is possible, without touching the source code, to Asterisk save the cdr with date in unix time instead of the default date. It's possible? Thanks in advance, -- Rodrigo Lang Opening your mind - Just another Open Source site http://openingyourmind.wordpress.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] Question about EuroBRI final 2 digits
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian
Gansberger
Sent: Thursday, February 10, 2011 5:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about EuroBRI final 2 digits
Hello,
Maybe try that:
In your incoming isdn context:
[isdn-incoming]
exten = s,1,Set(TIMEOUT(digits)=3)
exten = s,2,WaitExten(2)
exten = s,3,Dial(SIP/operator...)
exten = 10,1,Dial(SIP/10)
exten = 20,1,Dial(SIP/20)
So if a call comes in Asterisk waits, 2 seconds for further digits
dialed and if so jumps to the right extension in the context.
Overlapdial should be yes.
yours
christian gansberger
www.accm.at
On 3 February 2011 20:45, Cassius Smith cass...@cassius.org wrote:
Hello,
I have an installation in Austria; ISDN service provided by Austria
Telekom.
The main number of the service is 6 digits. Incoming calls may contain 2
additional digits, which I then use to route the call to the correct
extension. Telekom sends me all the digits.
My problem is that when someone tries to dial an 8 digit number to an
extension from an outside analog phone, AT sends the call before they
finish
dialing all 8 digits. Is there a way to prevent this, or to catch the
additional 2 digits somewhere in the stream? The receptionist is unhappy
because she gets all the 6-digit calls and must then transfer.
Is this a p2p vs p2mp issue?
Thanks in advance,
Cassius Smith
--
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You should be able to query the length of ${CALLERID(num)} and process the
full 8 digits that way. Telekom sends me all the digits tells me that the
number dialed to get to the extension arrives intact and that your dialplan
is truncating it.
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Re: [asterisk-users] Question about EuroBRI final 2 digits
On Thu, Feb 10, 2011 at 6:08 AM, Andrew Thomas a...@datavox.co.uk wrote: This sounds like a job for DISA. http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA to see if DISA helps. If OP is using Asterisk18, perhaps we should direct him to look here? https://wiki.asterisk.org/wiki/display/AST/Application_DISA cheers, -- -Bob -- _ -- 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] Question about EuroBRI final 2 digits
...or there :) Anyway AT sends the call before they finish dialling all 8 digits means that they don't send all the digits. Conflicting sentence in OP. Perhaps it would help if the OP could determine if AT actually send 6 or 8 digits in the signalling (I reckon it's 6). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Beers Sent: 10 February 2011 14:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about EuroBRI final 2 digits On Thu, Feb 10, 2011 at 6:08 AM, Andrew Thomas a...@datavox.co.uk wrote: This sounds like a job for DISA. http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA to see if DISA helps. If OP is using Asterisk18, perhaps we should direct him to look here? https://wiki.asterisk.org/wiki/display/AST/Application_DISA cheers, -- -Bob -- _ -- 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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] fail-over server
On Wed, Feb 9, 2011 at 6:55 AM, Vieri rentor...@yahoo.com wrote: [snip] Since all of the SIP devices in my LAN have static IP addresses, I can keep track of everyone on my own. For instance, could I do fake SIP registrations from localhost (the * server) and specify a LAN IP address? Have you looked at the 'defaultip' sip configuration option? Or setting host=IP for those devices? I would write a custom script that would execute whenever an Asterisk server takes over. As said earlier, this server would not have any SIP extensions registered at first and they would be registering slowly within 60 seconds or more. However, since I KNOW FOR SURE that some SIP devices are always online and have static IP addresses, can't I fool Asterisk by somehow registering via locahost but spoofing the source IP address? Maybe setting the source port to what it was exactly can be tougher but I *could* try to keep track of it. That sounds more complicated and likely to break than using Realtime. This way, whenever the Asterisk server that took over tries to bridge a call, it will try to connect to the fakely-registered IP address. I'm not using realtime for 2 reasons: 1- I'm using the FreePBX framework and there's no realtime backend unfortunately. Moving to Realtime and losing all the FreePBX goodies is time-consuming. Does anyone know how to use FreePBX + Realtime? This is unfortunate for most of the Asterisk GUI's available. 2- I don't have enough hardware resources to setup a server for the realtime DB that both Asterisk servers would connect to. Also, I wouldn't feel comfortable having just one DB server. For easier maintenance I would use a clustered database for realtime. However, I'm using Mysql 5.0 ndbcluster tables for other non-voip purposes and my experience hasn't been so great. I once had a power outage and all ndb table data was lost. Also, 5.0 ndb crashes in several occasions. As far as I can tell, it isn't reliable. I haven't tried 5.1 though. I have no experience with clustered postgresql. So run the DB on the same server as Asterisk, if your call volume allows it, and either replicate the data using the built-in DB replication or use DRBD between the two existing servers. We use DRBD between two Asterisk nodes on smaller installations for configurations and voicemail. It works very well for us. For MySQL Cluster to work well, the application has to be designed for it, and it is a RAM based storage. But that is a conversation for another list. -Jonathan -- _ -- 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] Early audio SIP sequence order question
On 02/10/2011 06:15 AM, Benoit Panizzon wrote: Hi Faisal http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Thank you, but I seem to miss the option which tells asterisk to pass audio even if no 183 or 200 is received. No, we don't set the 'r' Flag while dialing out. So, my question ist sill the same. Sould asterisk pass audio of it didn't yet receive a 183 or 200 message, or is the carrier doing wrong in sending early audio without 183? This does indeed sound like an Asterisk bug; Asterisk should be ready and willing to accept audio from the called SIP endpoint as soon as the INVITE is sent out with an SDP offer to receive audio. Now the real issue here may be the Dial() application not forwarding that audio to the caller, rather than Asterisk not 'accepting' the audio and turning it into internal media frames. The net result for you is the same, but the source of the problem is quite different. This can of course cause complications if Dial() is used to dial multiple endpoints... because then there could be multiple audio streams received from them as the call proceeds towards one of them answering. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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] Busy Detection on Analog Lines
Hi, I'm having an issue with busy detection, the busy is not being detected. Asterisk: 1.6.2.13 DAHDI: 2.4.0 Chandahdi: busydetect=yes, busycount=2 Indications zone = us, with the modifications for my country for busy: 425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off) I compiled with BUSY DETECT DEBUG. I can see: [Feb 10 15:48:06] DEBUG[26968]: dsp.c:1276 ast_dsp_busydetect: busy detector: FAILED with avgtone: 255, avgsilence 30 [Feb 10 15:48:06] DEBUG[26968]: dsp.c:1276 ast_dsp_busydetect: busy detector: FAILED with avgtone: 255, avgsilence 30 [Feb 10 15:48:06] DEBUG[26968]: dsp.c:1276 ast_dsp_busydetect: busy detector: FAILED with avgtone: 255, avgsilence 30 [Feb 10 15:48:40] DEBUG[26971]: dsp.c:1276 ast_dsp_busydetect: busy detector: FAILED with avgtone: 260, avgsilence 30 [Feb 10 15:48:40] DEBUG[26971]: dsp.c:1276 ast_dsp_busydetect: busy detector: FAILED with avgtone: 260, avgsilence 30 And when I hangup the line nothing more is shown of the busydetect debug, but the line is still on. Any ideas? Best 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] CDR with unix time.
On Thursday 10 February 2011 06:13:38 Rodrigo Lang wrote: I wonder if it is possible, without touching the source code, to Asterisk save the cdr with date in unix time instead of the default date. It's possible? The answer is, it depends upon the backend version you're using. With cdr_pgsql and cdr_mysql from 1.6.2 forward, if the column type is integer or float, then the unix timestamp will be used. -- Tilghman -- _ -- 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] fail-over server
--- On Thu, 2/10/11, Jonathan Thurman jonat...@thurmantech.com wrote: Have you looked at the 'defaultip' sip configuration option? Or setting host=IP for those devices? I've read that defaultip can only be used on type=peer and when host=dynamic. I use type=friend. host=IP seems to be OK for me. I actually tried this option some time ago but had trouble with something I can't recall right now so reverted to dynamic. I guess I'll have to give it another shot. I'll try that before migrating to realtime... Thanks Jonathan! -- _ -- 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] res_pgsql re-connect on db failure?
We are using PostgreSQL real-time connector (res_config_pgsql) with Asterisk 1.6.2.15. From time to time, we need to reset our PostgreSQL server, causing all active DB connections to close. While other packages in our system re-connect gracefully when this happens, Asterisk appears to not bother trying. It instead goes into an endless loop complaining that the connection has closed. Question -- is there any option I might be missing, to tell Asterisk to try re-connecting to PostgreSQL if the existing connection fails? Thank you, Bryan -- _ -- 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] Gtalk/Jabber Issue
OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can't seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes ;autoprune=no autoregister=yes [jb_jabber] type=client serverhost=talk.google.com username=xx...@gmail.com/Talk secret=XXX port=5222 usetls=yes usesasl=yes ;status=Available statusmessage=Connected via Asterisk ;timeout=100 ;keepalive=yes Extensions.conf [google-in] exten = s,1,NoOp(Call from GTalk) exten = s,n,Set(CallerID(Name)=From GoogleTalk) exten = s,n,Dial(SIP/1000) jabber show connected Jabber Users and their status: User: xxx...@gmail.com/Talk - Connected Number of users: 1 CLI on incoming Call bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800E94 type=setses:session type=initiate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0 name=PCMU clockrate=8000/pho:payload-type id=101 name=telephone-event//pho:descriptiontransport behind-symmetric-nat=false can-receive-from-symmetric-nat=false xmlns=http://www.google.com/transport/raw-udp/transport xmlns=http://www.google.com/transport/p2p//ses:session/iq bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800EB9 type=setses:session type=terminate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:call-ended xmlns:pho=http://www.google.com/session/phone;Call cancelled/pho:call-ended/ses:session/iq bannana*CLI it doesn't even try to fire the google-in context ? Lastest Version of iksemel Installed, asterisk was rebuild after installed, asterisk sees both jabber/gtalk commands. It just will NOT ring my dialplan. -- _ -- 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] Gtalk/Jabber Issue
Sorry, Asterisk Build 1.6.2.7 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Thursday, February 10, 2011 6:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Gtalk/Jabber Issue OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can't seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes ;autoprune=no autoregister=yes [jb_jabber] type=client serverhost=talk.google.com username=xx...@gmail.com/Talk secret=XXX port=5222 usetls=yes usesasl=yes ;status=Available statusmessage=Connected via Asterisk ;timeout=100 ;keepalive=yes Extensions.conf [google-in] exten = s,1,NoOp(Call from GTalk) exten = s,n,Set(CallerID(Name)=From GoogleTalk) exten = s,n,Dial(SIP/1000) jabber show connected Jabber Users and their status: User: xxx...@gmail.com/Talk - Connected Number of users: 1 CLI on incoming Call bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800E94 type=setses:session type=initiate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0 name=PCMU clockrate=8000/pho:payload-type id=101 name=telephone-event//pho:descriptiontransport behind-symmetric-nat=false can-receive-from-symmetric-nat=false xmlns=http://www.google.com/transport/raw-udp/transport xmlns=http://www.google.com/transport/p2p//ses:session/iq bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800EB9 type=setses:session type=terminate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:call-ended xmlns:pho=http://www.google.com/session/phone;Call cancelled/pho:call-ended/ses:session/iq bannana*CLI it doesn't even try to fire the google-in context ? Lastest Version of iksemel Installed, asterisk was rebuild after installed, asterisk sees both jabber/gtalk commands. It just will NOT ring my dialplan. -- _ -- 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] meetme conference playback of random sound file
i have been trying to find a way to accomplish the following but have not found a method in which to do so: i am trying to configure the meetme conference (asterisk 1.8) to play a * random* sound file from a specific directory prior to it dropping the caller into the conference itself. i am able to successfully get it to play a specific file prior to entering the conference unsure how to implement this sort of randomization. Is this possible? Any help will be greatly appreciated. john jolly jgjolly[at]gmail.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] meetme conference playback of random sound file
On Thu, Feb 10, 2011 at 04:58:05PM -0800, John Jolly wrote: i am trying to configure the meetme conference (asterisk 1.8) to play a * random* sound file from a specific directory prior to it dropping the caller into the conference itself. Absent an Asterisk-specific solution, how about a separate process which would link a random file into a fixed pathname? (Fired off from cron, perhaps.) Roger -- _ -- 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] meetme conference playback of random sound file
On Thu, 10 Feb 2011, John Jolly wrote: i am trying to configure the meetme conference (asterisk 1.8) to play a random sound file from a specific directory prior to it dropping the caller into the conference itself. i am able to successfully get it to play a specific file prior to entering the conference unsure how to implement this sort of randomization. Who is the sound file played to? The caller or the conference? Please show what you are using now. Would an AGI that selected a random file from the directory and set the path as a channel variable work? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] Gtalk/Jabber Issue
You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in the other. Thanks, --Warren Selby, dCAP On Feb 10, 2011, at 5:55 PM, William Stillwell will...@stillwellsoft.com wrote: Sorry, Asterisk Build 1.6.2.7 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Thursday, February 10, 2011 6:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Gtalk/Jabber Issue OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can’t seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes ;autoprune=no autoregister=yes [jb_jabber] type=client serverhost=talk.google.com username=xx...@gmail.com/Talk secret=XXX port=5222 usetls=yes usesasl=yes ;status=Available statusmessage=Connected via Asterisk ;timeout=100 ;keepalive=yes Extensions.conf [google-in] exten = s,1,NoOp(Call from GTalk) exten = s,n,Set(CallerID(Name)=From GoogleTalk) exten = s,n,Dial(SIP/1000) jabber show connected Jabber Users and their status: User: xxx...@gmail.com/Talk - Connected Number of users: 1 CLI on incoming Call bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800E94 type=setses:session type=initiate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0 name=PCMU clockrate=8000/pho:payload-type id=101 name=telephone-event//pho:descriptiontransport behind-symmetric-nat=false can-receive-from-symmetric-nat=false xmlns=http://www.google.com/transport/raw-udp/transport xmlns=http://www.google.com/transport/p2p//ses:session/iq bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800EB9 type=setses:session type=terminate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:call-ended xmlns:pho=http://www.google.com/session/phone;Call cancelled/pho:call-ended/ses:session/iq bannana*CLI it doesn’t even try to fire the google-in context ? Lastest Version of iksemel Installed, asterisk was rebuild after installed, asterisk sees both jabber/gtalk commands. It just will NOT ring my dialplan. -- _ -- 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] Gtalk/Jabber Issue
Yeah, that was a typo, but I fixed, still no dice. The incoming jabber call doesn’t fire the gtalk connection. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Thursday, February 10, 2011 10:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in the other. Thanks, --Warren Selby, dCAP On Feb 10, 2011, at 5:55 PM, William Stillwell will...@stillwellsoft.com wrote: Sorry, Asterisk Build 1.6.2.7 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Thursday, February 10, 2011 6:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Gtalk/Jabber Issue OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can’t seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes ;autoprune=no autoregister=yes [jb_jabber] type=client serverhost=talk.google.com username=xx...@gmail.com/Talk secret=XXX port=5222 usetls=yes usesasl=yes ;status=Available statusmessage=Connected via Asterisk ;timeout=100 ;keepalive=yes Extensions.conf [google-in] exten = s,1,NoOp(Call from GTalk) exten = s,n,Set(CallerID(Name)=From GoogleTalk) exten = s,n,Dial(SIP/1000) jabber show connected Jabber Users and their status: User: xxx...@gmail.com/Talk - Connected Number of users: 1 CLI on incoming Call bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800E94 type=setses:session type=initiate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0 name=PCMU clockrate=8000/pho:payload-type id=101 name=telephone-event//pho:descriptiontransport behind-symmetric-nat=false can-receive-from-symmetric-nat=false xmlns=http://www.google.com/transport/raw-udp/transport xmlns=http://www.google.com/transport/p2p//ses:session/iq bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800EB9 type=setses:session type=terminate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:call-ended xmlns:pho=http://www.google.com/session/phone;Call cancelled/pho:call-ended/ses:session/iq bannana*CLI it doesn’t even try to fire the google-in context ? Lastest Version of iksemel Installed, asterisk was rebuild after installed, asterisk sees both jabber/gtalk commands. It just will NOT ring my dialplan. -- _ -- 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] Gtalk/Jabber Issue
Ok, stopped asterisk Backed up all modules Recompiled asterisk to lastest version. Same thing… jabber call come in, but no firing of the gtalk/extension.. Now running build 1.6.2.16.1 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Thursday, February 10, 2011 11:44 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Gtalk/Jabber Issue Yeah, that was a typo, but I fixed, still no dice. The incoming jabber call doesn’t fire the gtalk connection. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Thursday, February 10, 2011 10:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in the other. Thanks, --Warren Selby, dCAP On Feb 10, 2011, at 5:55 PM, William Stillwell will...@stillwellsoft.com wrote: Sorry, Asterisk Build 1.6.2.7 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Thursday, February 10, 2011 6:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Gtalk/Jabber Issue OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can’t seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes ;autoprune=no autoregister=yes [jb_jabber] type=client serverhost=talk.google.com username=xx...@gmail.com/Talk secret=XXX port=5222 usetls=yes usesasl=yes ;status=Available statusmessage=Connected via Asterisk ;timeout=100 ;keepalive=yes Extensions.conf [google-in] exten = s,1,NoOp(Call from GTalk) exten = s,n,Set(CallerID(Name)=From GoogleTalk) exten = s,n,Dial(SIP/1000) jabber show connected Jabber Users and their status: User: xxx...@gmail.com/Talk - Connected Number of users: 1 CLI on incoming Call bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800E94 type=setses:session type=initiate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0 name=PCMU clockrate=8000/pho:payload-type id=101 name=telephone-event//pho:descriptiontransport behind-symmetric-nat=false can-receive-from-symmetric-nat=false xmlns=http://www.google.com/transport/raw-udp/transport xmlns=http://www.google.com/transport/p2p//ses:session/iq bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800EB9 type=setses:session type=terminate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:call-ended xmlns:pho=http://www.google.com/session/phone;Call cancelled/pho:call-ended/ses:session/iq bannana*CLI it doesn’t even try to fire the google-in context ? Lastest Version of iksemel Installed, asterisk was rebuild after installed, asterisk sees both jabber/gtalk commands. It just will NOT ring my dialplan. -- _ -- 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] Gtalk/Jabber Issue
On Thu, Feb 10, 2011 at 11:17 PM, William Stillwell will...@stillwellsoft.com wrote: Ok, stopped asterisk Backed up all modules Recompiled asterisk to lastest version. Same thing… jabber call come in, but no firing of the gtalk/extension.. Now running build 1.6.2.16.1 Try adding the following to your [google-in] context in extension.conf: exten = _.,1,Verbose(Call from GTalk - catchall) exten = _.,n,Set(CallerID(Name)=From GoogleTalk) exten = _.,n,Dial(SIP/1000) -- Thanks, --Warren Selby, dCAP http://www.selbytech.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] Gtalk/Jabber Issue
William, Have you tried outgoing calls? What happens there? Have you restarted the Asterisk after you fixed the typo? -Vladimir On 2/10/2011 10:44 PM, William Stillwell wrote: Yeah, that was a typo, but I fixed, still no dice. The incoming jabber call doesn’t fire the gtalk connection. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Warren Selby *Sent:* Thursday, February 10, 2011 10:16 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in the other. Thanks, --Warren Selby, dCAP On Feb 10, 2011, at 5:55 PM, William Stillwell will...@stillwellsoft.com mailto:will...@stillwellsoft.com wrote: Sorry, Asterisk Build 1.6.2.7 *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *William Stillwell *Sent:* Thursday, February 10, 2011 6:50 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Gtalk/Jabber Issue OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can’t seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes ;autoprune=no autoregister=yes [jb_jabber] type=client serverhost=talk.google.com username=xx...@gmail.com mailto:username=xx...@gmail.com/Talk secret=XXX port=5222 usetls=yes usesasl=yes ;status=Available statusmessage=Connected via Asterisk ;timeout=100 ;keepalive=yes Extensions.conf [google-in] exten = s,1,NoOp(Call from GTalk) exten = s,n,Set(CallerID(Name)=From GoogleTalk) exten = s,n,Dial(SIP/1000) jabber show connected Jabber Users and their status: User: xxx...@gmail.com mailto:xxx...@gmail.com/Talk - Connected Number of users: 1 CLI on incoming Call bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 mailto:+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 mailto:**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800E94 type=setses:session type=initiate id=SIP1007753261@10.218.122.83 mailto:SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 mailto:+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0 name=PCMU clockrate=8000/pho:payload-type id=101 name=telephone-event//pho:descriptiontransport behind-symmetric-nat=false can-receive-from-symmetric-nat=false xmlns=http://www.google.com/transport/raw-udp/transport xmlns=http://www.google.com/transport/p2p//ses:session/iq bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 mailto:+1@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 mailto:**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800EB9 type=setses:session type=terminate id=SIP1007753261@10.218.122.83 mailto:SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 mailto:+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:call-ended xmlns:pho=http://www.google.com/session/phone;Call cancelled/pho:call-ended/ses:session/iq bannana*CLI it doesn’t even try to fire the google-in context ? Lastest Version of iksemel Installed, asterisk was rebuild after installed, asterisk sees both jabber/gtalk commands. It just will NOT ring my dialplan. -- _ -- 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
Re: [asterisk-users] Gtalk/Jabber Issue
William, I have just noticed that you have several configuration statements commented out. I would suggest to un-comment the status= in jabber.conf. I would also suggest to un-comment the timeout=, I am not that concerned of the keepalive=. You can reload jabber, no need to restart the Asterisk. -Vladimir On 2/10/2011 11:40 PM, Vladimir Mikhelson wrote: William, Have you tried outgoing calls? What happens there? Have you restarted the Asterisk after you fixed the typo? -Vladimir On 2/10/2011 10:44 PM, William Stillwell wrote: Yeah, that was a typo, but I fixed, still no dice. The incoming jabber call doesn’t fire the gtalk connection. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Warren Selby *Sent:* Thursday, February 10, 2011 10:16 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in the other. Thanks, --Warren Selby, dCAP On Feb 10, 2011, at 5:55 PM, William Stillwell will...@stillwellsoft.com mailto:will...@stillwellsoft.com wrote: Sorry, Asterisk Build 1.6.2.7 *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *William Stillwell *Sent:* Thursday, February 10, 2011 6:50 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Gtalk/Jabber Issue OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can’t seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes ;autoprune=no autoregister=yes [jb_jabber] type=client serverhost=talk.google.com username=xx...@gmail.com mailto:username=xx...@gmail.com/Talk secret=XXX port=5222 usetls=yes usesasl=yes ;status=Available statusmessage=Connected via Asterisk ;timeout=100 ;keepalive=yes Extensions.conf [google-in] exten = s,1,NoOp(Call from GTalk) exten = s,n,Set(CallerID(Name)=From GoogleTalk) exten = s,n,Dial(SIP/1000) jabber show connected Jabber Users and their status: User: xxx...@gmail.com mailto:xxx...@gmail.com/Talk - Connected Number of users: 1 CLI on incoming Call bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 mailto:+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 mailto:**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800E94 type=setses:session type=initiate id=SIP1007753261@10.218.122.83 mailto:SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 mailto:+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0 name=PCMU clockrate=8000/pho:payload-type id=101 name=telephone-event//pho:descriptiontransport behind-symmetric-nat=false can-receive-from-symmetric-nat=false xmlns=http://www.google.com/transport/raw-udp/transport xmlns=http://www.google.com/transport/p2p//ses:session/iq bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 mailto:+1@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 mailto:**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800EB9 type=setses:session type=terminate id=SIP1007753261@10.218.122.83 mailto:SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 mailto:+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:call-ended xmlns:pho=http://www.google.com/session/phone;Call cancelled/pho:call-ended/ses:session/iq bannana*CLI it doesn’t even try to fire the google-in context ? Lastest Version of iksemel Installed, asterisk was rebuild after installed, asterisk sees both jabber/gtalk commands.
Re: [asterisk-users] Gtalk/Jabber Issue
I was getting unable to make channel.. So, this is what I am doing.. Service stop asterisk Purge modules Make clean Remove all traces of iskemel Recompile that. With , add needed entrée into ldconfig. Verify iksemel loaded via ldconfig –p | grep semel. Change to /asterisk source location Make clean Now ./configure asterisk Make menuselect, make sure chan_gtalk, and res_jabber as selected. Make Make install Start asterisk.. Trying inbound.. Same thing, jabber call comes in, doesn’t fire the gtalk extension.. Outbound call , I get: [Feb 11 00:52:18] ERROR[440]: chan_gtalk.c:934 gtalk_alloc: no gtalk capable clients to talk to. [Feb 11 00:52:18] WARNING[440]: app_dial.c:1759 dial_exec_full: Unable to create channel of type 'Gtalk' (cause 0 - Unknown) ?? jabber/jingle/gtalk cmd all exist, and modules loaded. From: Vladimir Mikhelson [mailto:v...@mikhelson.com] Sent: Friday, February 11, 2011 12:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: William Stillwell Subject: Re: [asterisk-users] Gtalk/Jabber Issue William, Have you tried outgoing calls? What happens there? Have you restarted the Asterisk after you fixed the typo? -Vladimir On 2/10/2011 10:44 PM, William Stillwell wrote: Yeah, that was a typo, but I fixed, still no dice. The incoming jabber call doesn’t fire the gtalk connection. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Thursday, February 10, 2011 10:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in the other. Thanks, --Warren Selby, dCAP On Feb 10, 2011, at 5:55 PM, William Stillwell will...@stillwellsoft.com wrote: Sorry, Asterisk Build 1.6.2.7 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Thursday, February 10, 2011 6:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Gtalk/Jabber Issue OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can’t seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes ;autoprune=no autoregister=yes [jb_jabber] type=client serverhost=talk.google.com username=xx...@gmail.com/Talk secret=XXX port=5222 usetls=yes usesasl=yes ;status=Available statusmessage=Connected via Asterisk ;timeout=100 ;keepalive=yes Extensions.conf [google-in] exten = s,1,NoOp(Call from GTalk) exten = s,n,Set(CallerID(Name)=From GoogleTalk) exten = s,n,Dial(SIP/1000) jabber show connected Jabber Users and their status: User: xxx...@gmail.com/Talk - Connected Number of users: 1 CLI on incoming Call bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800E94 type=setses:session type=initiate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0 name=PCMU clockrate=8000/pho:payload-type id=101 name=telephone-event//pho:descriptiontransport behind-symmetric-nat=false can-receive-from-symmetric-nat=false xmlns=http://www.google.com/transport/raw-udp/transport xmlns=http://www.google.com/transport/p2p//ses:session/iq bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800EB9 type=setses:session type=terminate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:call-ended xmlns:pho=http://www.google.com/session/phone;Call cancelled/pho:call-ended/ses:session/iq bannana*CLI it doesn’t even try to fire the google-in context ? Lastest Version of iksemel Installed, asterisk was rebuild after installed, asterisk sees both jabber/gtalk commands. It just will NOT ring my dialplan. -- _ -- 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:
Re: [asterisk-users] Gtalk/Jabber Issue
Still no dice.. This make no since.. ive gone over the config a million times now.. The windows gtalk /voice client works just fine. (incoming and outgoing calls) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Friday, February 11, 2011 12:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue William, I have just noticed that you have several configuration statements commented out. I would suggest to un-comment the status= in jabber.conf. I would also suggest to un-comment the timeout=, I am not that concerned of the keepalive=. You can reload jabber, no need to restart the Asterisk. -Vladimir On 2/10/2011 11:40 PM, Vladimir Mikhelson wrote: William, Have you tried outgoing calls? What happens there? Have you restarted the Asterisk after you fixed the typo? -Vladimir On 2/10/2011 10:44 PM, William Stillwell wrote: Yeah, that was a typo, but I fixed, still no dice. The incoming jabber call doesn’t fire the gtalk connection. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Thursday, February 10, 2011 10:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in the other. Thanks, --Warren Selby, dCAP On Feb 10, 2011, at 5:55 PM, William Stillwell will...@stillwellsoft.com wrote: Sorry, Asterisk Build 1.6.2.7 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Thursday, February 10, 2011 6:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Gtalk/Jabber Issue OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can’t seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes ;autoprune=no autoregister=yes [jb_jabber] type=client serverhost=talk.google.com username=xx...@gmail.com/Talk secret=XXX port=5222 usetls=yes usesasl=yes ;status=Available statusmessage=Connected via Asterisk ;timeout=100 ;keepalive=yes Extensions.conf [google-in] exten = s,1,NoOp(Call from GTalk) exten = s,n,Set(CallerID(Name)=From GoogleTalk) exten = s,n,Dial(SIP/1000) jabber show connected Jabber Users and their status: User: xxx...@gmail.com/Talk - Connected Number of users: 1 CLI on incoming Call bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800E94 type=setses:session type=initiate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0 name=PCMU clockrate=8000/pho:payload-type id=101 name=telephone-event//pho:descriptiontransport behind-symmetric-nat=false can-receive-from-symmetric-nat=false xmlns=http://www.google.com/transport/raw-udp/transport xmlns=http://www.google.com/transport/p2p//ses:session/iq bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800EB9 type=setses:session type=terminate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:call-ended xmlns:pho=http://www.google.com/session/phone;Call cancelled/pho:call-ended/ses:session/iq bannana*CLI it doesn’t even try to fire the google-in context ? Lastest Version of iksemel Installed, asterisk was rebuild after installed, asterisk sees both jabber/gtalk commands. It just will NOT ring my dialplan. -- _ -- 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:
[asterisk-users] sangoma wanpipe install error
Trying to install wanpipe 3.5.18. No errors during compile. But when I reach the point where wanpipe and dahdi_cfg is started, I encountered an error. Starting WAN Router... Loading WAN drivers: wanpipe done. Starting up device: wanpipe1 wanconfig: WAN device wanpipe1 driver load failed !! : ioctl(wanpipe1,ROUTER_SETUP) failed: : 22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errors Configuring interfaces: w1g1 w1g1: ERROR while getting interface flags: No such device done. /etc/wanpipe/scripts/start: 7: Syntax error: Bad for loop variable DAHDI_SPANCONFIG failed on span 1: No such device or address (6) Dont know why I still keep getting a 'No such device' error even if the device was detected (Sangoma a104de, setup asked to configure/skip the 4 ports) before the error happened. -- _ -- 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] Gtalk/Jabber Issue
William, I have gone through the similar frustration recently. Everything works as of early morning yesterday. The big difference, I am on 1.8.2.3. Have you seen this ticket on the tracker https://issues.asterisk.org/view.php?id=10512 ? Anything applicable to your case? The messages are identical to yours on the outgoing call. -Vladimir On 2/11/2011 12:32 AM, William Stillwell wrote: Still no dice.. This make no since.. ive gone over the config a million times now.. The windows gtalk /voice client works just fine. (incoming and outgoing calls) *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Vladimir Mikhelson *Sent:* Friday, February 11, 2011 12:51 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue William, I have just noticed that you have several configuration statements commented out. I would suggest to un-comment the status= in jabber.conf. I would also suggest to un-comment the timeout=, I am not that concerned of the keepalive=. You can reload jabber, no need to restart the Asterisk. -Vladimir On 2/10/2011 11:40 PM, Vladimir Mikhelson wrote: William, Have you tried outgoing calls? What happens there? Have you restarted the Asterisk after you fixed the typo? -Vladimir On 2/10/2011 10:44 PM, William Stillwell wrote: Yeah, that was a typo, but I fixed, still no dice. The incoming jabber call doesn’t fire the gtalk connection. *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Warren Selby *Sent:* Thursday, February 10, 2011 10:16 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in the other. Thanks, --Warren Selby, dCAP On Feb 10, 2011, at 5:55 PM, William Stillwell will...@stillwellsoft.com mailto:will...@stillwellsoft.com wrote: Sorry, Asterisk Build 1.6.2.7 *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *William Stillwell *Sent:* Thursday, February 10, 2011 6:50 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Gtalk/Jabber Issue OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can’t seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes ;autoprune=no autoregister=yes [jb_jabber] type=client serverhost=talk.google.com username=xx...@gmail.com mailto:username=xx...@gmail.com/Talk secret=XXX port=5222 usetls=yes usesasl=yes ;status=Available statusmessage=Connected via Asterisk ;timeout=100 ;keepalive=yes Extensions.conf [google-in] exten = s,1,NoOp(Call from GTalk) exten = s,n,Set(CallerID(Name)=From GoogleTalk) exten = s,n,Dial(SIP/1000) jabber show connected Jabber Users and their status: User: xxx...@gmail.com mailto:xxx...@gmail.com/Talk - Connected Number of users: 1 CLI on incoming Call bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 mailto:+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 mailto:**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800E94 type=setses:session type=initiate id=SIP1007753261@10.218.122.83 mailto:SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 mailto:+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0 name=PCMU clockrate=8000/pho:payload-type id=101 name=telephone-event//pho:descriptiontransport behind-symmetric-nat=false can-receive-from-symmetric-nat=false xmlns=http://www.google.com/transport/raw-udp/transport xmlns=http://www.google.com/transport/p2p//ses:session/iq bannana*CLI JABBER:
Re: [asterisk-users] Gtalk/Jabber Issue
William, Another thing. Have you tried calling from GMail? If not please make sure you can send/receive calls there. One more test. Go to your GV Account Settings / Phones, Edit Google Chat, Save Watch for the pink error messages in the upper portion of the screen. -Vladimir On 2/11/2011 12:32 AM, William Stillwell wrote: Still no dice.. This make no since.. ive gone over the config a million times now.. The windows gtalk /voice client works just fine. (incoming and outgoing calls) *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Vladimir Mikhelson *Sent:* Friday, February 11, 2011 12:51 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue William, I have just noticed that you have several configuration statements commented out. I would suggest to un-comment the status= in jabber.conf. I would also suggest to un-comment the timeout=, I am not that concerned of the keepalive=. You can reload jabber, no need to restart the Asterisk. -Vladimir On 2/10/2011 11:40 PM, Vladimir Mikhelson wrote: William, Have you tried outgoing calls? What happens there? Have you restarted the Asterisk after you fixed the typo? -Vladimir On 2/10/2011 10:44 PM, William Stillwell wrote: Yeah, that was a typo, but I fixed, still no dice. The incoming jabber call doesn’t fire the gtalk connection. *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Warren Selby *Sent:* Thursday, February 10, 2011 10:16 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in the other. Thanks, --Warren Selby, dCAP On Feb 10, 2011, at 5:55 PM, William Stillwell will...@stillwellsoft.com mailto:will...@stillwellsoft.com wrote: Sorry, Asterisk Build 1.6.2.7 *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *William Stillwell *Sent:* Thursday, February 10, 2011 6:50 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Gtalk/Jabber Issue OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can’t seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes ;autoprune=no autoregister=yes [jb_jabber] type=client serverhost=talk.google.com username=xx...@gmail.com mailto:username=xx...@gmail.com/Talk secret=XXX port=5222 usetls=yes usesasl=yes ;status=Available statusmessage=Connected via Asterisk ;timeout=100 ;keepalive=yes Extensions.conf [google-in] exten = s,1,NoOp(Call from GTalk) exten = s,n,Set(CallerID(Name)=From GoogleTalk) exten = s,n,Dial(SIP/1000) jabber show connected Jabber Users and their status: User: xxx...@gmail.com mailto:xxx...@gmail.com/Talk - Connected Number of users: 1 CLI on incoming Call bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 mailto:+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 mailto:**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800E94 type=setses:session type=initiate id=SIP1007753261@10.218.122.83 mailto:SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 mailto:+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0 name=PCMU clockrate=8000/pho:payload-type id=101 name=telephone-event//pho:descriptiontransport behind-symmetric-nat=false can-receive-from-symmetric-nat=false xmlns=http://www.google.com/transport/raw-udp/transport xmlns=http://www.google.com/transport/p2p//ses:session/iq bannana*CLI JABBER: jb_jabber INCOMING: iq
Re: [asterisk-users] Gtalk/Jabber Issue
I don’t’ appear to have an jabber [] OUTGOING packets? I get just 1 incoming packet, and it just sits there, until it rings to voicemail. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Friday, February 11, 2011 1:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue William, I have gone through the similar frustration recently. Everything works as of early morning yesterday. The big difference, I am on 1.8.2.3. Have you seen this ticket on the tracker https://issues.asterisk.org/view.php?id=10512 ? Anything applicable to your case? The messages are identical to yours on the outgoing call. -Vladimir On 2/11/2011 12:32 AM, William Stillwell wrote: Still no dice.. This make no since.. ive gone over the config a million times now.. The windows gtalk /voice client works just fine. (incoming and outgoing calls) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Friday, February 11, 2011 12:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue William, I have just noticed that you have several configuration statements commented out. I would suggest to un-comment the status= in jabber.conf. I would also suggest to un-comment the timeout=, I am not that concerned of the keepalive=. You can reload jabber, no need to restart the Asterisk. -Vladimir On 2/10/2011 11:40 PM, Vladimir Mikhelson wrote: William, Have you tried outgoing calls? What happens there? Have you restarted the Asterisk after you fixed the typo? -Vladimir On 2/10/2011 10:44 PM, William Stillwell wrote: Yeah, that was a typo, but I fixed, still no dice. The incoming jabber call doesn’t fire the gtalk connection. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Thursday, February 10, 2011 10:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in the other. Thanks, --Warren Selby, dCAP On Feb 10, 2011, at 5:55 PM, William Stillwell will...@stillwellsoft.com wrote: Sorry, Asterisk Build 1.6.2.7 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Thursday, February 10, 2011 6:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Gtalk/Jabber Issue OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can’t seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes ;autoprune=no autoregister=yes [jb_jabber] type=client serverhost=talk.google.com username=xx...@gmail.com/Talk secret=XXX port=5222 usetls=yes usesasl=yes ;status=Available statusmessage=Connected via Asterisk ;timeout=100 ;keepalive=yes Extensions.conf [google-in] exten = s,1,NoOp(Call from GTalk) exten = s,n,Set(CallerID(Name)=From GoogleTalk) exten = s,n,Dial(SIP/1000) jabber show connected Jabber Users and their status: User: xxx...@gmail.com/Talk - Connected Number of users: 1 CLI on incoming Call bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800E94 type=setses:session type=initiate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0 name=PCMU clockrate=8000/pho:payload-type id=101 name=telephone-event//pho:descriptiontransport behind-symmetric-nat=false can-receive-from-symmetric-nat=false xmlns=http://www.google.com/transport/raw-udp/transport xmlns=http://www.google.com/transport/p2p//ses:session/iq bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800EB9 type=setses:session type=terminate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:call-ended xmlns:pho=http://www.google.com/session/phone;Call cancelled/pho:call-ended/ses:session/iq bannana*CLI it
Re: [asterisk-users] Gtalk/Jabber Issue
William, Another thing to exclude is networking. Can you verify that nothing blocks the specific traffic on your network? Any chance of taking the packet trace on your gateway? -Vladimir On 2/11/2011 1:18 AM, William Stillwell wrote: I don’t’ appear to have an jabber [] OUTGOING packets? I get just 1 incoming packet, and it just sits there, until it rings to voicemail. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Vladimir Mikhelson *Sent:* Friday, February 11, 2011 1:47 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue William, I have gone through the similar frustration recently. Everything works as of early morning yesterday. The big difference, I am on 1.8.2.3. Have you seen this ticket on the tracker https://issues.asterisk.org/view.php?id=10512 ? Anything applicable to your case? The messages are identical to yours on the outgoing call. -Vladimir On 2/11/2011 12:32 AM, William Stillwell wrote: Still no dice.. This make no since.. ive gone over the config a million times now.. The windows gtalk /voice client works just fine. (incoming and outgoing calls) *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Vladimir Mikhelson *Sent:* Friday, February 11, 2011 12:51 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue William, I have just noticed that you have several configuration statements commented out. I would suggest to un-comment the status= in jabber.conf. I would also suggest to un-comment the timeout=, I am not that concerned of the keepalive=. You can reload jabber, no need to restart the Asterisk. -Vladimir On 2/10/2011 11:40 PM, Vladimir Mikhelson wrote: William, Have you tried outgoing calls? What happens there? Have you restarted the Asterisk after you fixed the typo? -Vladimir On 2/10/2011 10:44 PM, William Stillwell wrote: Yeah, that was a typo, but I fixed, still no dice. The incoming jabber call doesn’t fire the gtalk connection. *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Warren Selby *Sent:* Thursday, February 10, 2011 10:16 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in the other. Thanks, --Warren Selby, dCAP On Feb 10, 2011, at 5:55 PM, William Stillwell will...@stillwellsoft.com mailto:will...@stillwellsoft.com wrote: Sorry, Asterisk Build 1.6.2.7 *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *William Stillwell *Sent:* Thursday, February 10, 2011 6:50 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Gtalk/Jabber Issue OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can’t seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes ;autoprune=no autoregister=yes [jb_jabber] type=client serverhost=talk.google.com username=xx...@gmail.com mailto:username=xx...@gmail.com/Talk secret=XXX port=5222 usetls=yes usesasl=yes ;status=Available statusmessage=Connected via Asterisk ;timeout=100 ;keepalive=yes Extensions.conf [google-in] exten = s,1,NoOp(Call from GTalk) exten = s,n,Set(CallerID(Name)=From GoogleTalk) exten = s,n,Dial(SIP/1000) jabber show connected Jabber Users and their status: User: xxx...@gmail.com mailto:xxx...@gmail.com/Talk - Connected Number of users: 1 CLI on incoming Call bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 mailto:+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 mailto:**@gmail.com/TalkD876FAA0
Re: [asterisk-users] Gtalk/Jabber Issue
I only have one gtalk account. I Double checked the chat settings. For some reason jabber is not sending any outbound response packets at all.. not sure why. Will need to see if I can stuff some more debug code into res_jabber.c and figure out whats going on, debug seems limited. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Friday, February 11, 2011 1:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue William, Another thing. Have you tried calling from GMail? If not please make sure you can send/receive calls there. One more test. Go to your GV Account Settings / Phones, Edit Google Chat, Save Watch for the pink error messages in the upper portion of the screen. -Vladimir On 2/11/2011 12:32 AM, William Stillwell wrote: Still no dice.. This make no since.. ive gone over the config a million times now.. The windows gtalk /voice client works just fine. (incoming and outgoing calls) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Friday, February 11, 2011 12:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue William, I have just noticed that you have several configuration statements commented out. I would suggest to un-comment the status= in jabber.conf. I would also suggest to un-comment the timeout=, I am not that concerned of the keepalive=. You can reload jabber, no need to restart the Asterisk. -Vladimir On 2/10/2011 11:40 PM, Vladimir Mikhelson wrote: William, Have you tried outgoing calls? What happens there? Have you restarted the Asterisk after you fixed the typo? -Vladimir On 2/10/2011 10:44 PM, William Stillwell wrote: Yeah, that was a typo, but I fixed, still no dice. The incoming jabber call doesn’t fire the gtalk connection. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Thursday, February 10, 2011 10:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in the other. Thanks, --Warren Selby, dCAP On Feb 10, 2011, at 5:55 PM, William Stillwell will...@stillwellsoft.com wrote: Sorry, Asterisk Build 1.6.2.7 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Thursday, February 10, 2011 6:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Gtalk/Jabber Issue OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can’t seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes ;autoprune=no autoregister=yes [jb_jabber] type=client serverhost=talk.google.com username=xx...@gmail.com/Talk secret=XXX port=5222 usetls=yes usesasl=yes ;status=Available statusmessage=Connected via Asterisk ;timeout=100 ;keepalive=yes Extensions.conf [google-in] exten = s,1,NoOp(Call from GTalk) exten = s,n,Set(CallerID(Name)=From GoogleTalk) exten = s,n,Dial(SIP/1000) jabber show connected Jabber Users and their status: User: xxx...@gmail.com/Talk - Connected Number of users: 1 CLI on incoming Call bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800E94 type=setses:session type=initiate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0 name=PCMU clockrate=8000/pho:payload-type id=101 name=telephone-event//pho:descriptiontransport behind-symmetric-nat=false can-receive-from-symmetric-nat=false xmlns=http://www.google.com/transport/raw-udp/transport xmlns=http://www.google.com/transport/p2p//ses:session/iq bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800EB9 type=setses:session type=terminate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:call-ended
Re: [asterisk-users] Gtalk/Jabber Issue
Hi William, just to know that gtalk/asterisk works in your environment you could quickly create a virtual server and install an asterisk 1.8 with this guide http://highsecurity.blogspot.com/2010/11/googlevoice-asterisk-18-with-freepbx.html which works fine for me. this way you know for sure that it really works and now it is sth to do with asterisk version/configs/dial plan. On Fri, Feb 11, 2011 at 3:41 PM, Vladimir Mikhelson v...@mikhelson.com wrote: William, Another thing to exclude is networking. Can you verify that nothing blocks the specific traffic on your network? Any chance of taking the packet trace on your gateway? -Vladimir On 2/11/2011 1:18 AM, William Stillwell wrote: I don’t’ appear to have an jabber [] OUTGOING packets? I get just 1 incoming packet, and it just sits there, until it rings to voicemail. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Friday, February 11, 2011 1:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue William, I have gone through the similar frustration recently. Everything works as of early morning yesterday. The big difference, I am on 1.8.2.3. Have you seen this ticket on the tracker https://issues.asterisk.org/view.php?id=10512 ? Anything applicable to your case? The messages are identical to yours on the outgoing call. -Vladimir On 2/11/2011 12:32 AM, William Stillwell wrote: Still no dice.. This make no since.. ive gone over the config a million times now.. The windows gtalk /voice client works just fine. (incoming and outgoing calls) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Friday, February 11, 2011 12:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue William, I have just noticed that you have several configuration statements commented out. I would suggest to un-comment the status= in jabber.conf. I would also suggest to un-comment the timeout=, I am not that concerned of the keepalive=. You can reload jabber, no need to restart the Asterisk. -Vladimir On 2/10/2011 11:40 PM, Vladimir Mikhelson wrote: William, Have you tried outgoing calls? What happens there? Have you restarted the Asterisk after you fixed the typo? -Vladimir On 2/10/2011 10:44 PM, William Stillwell wrote: Yeah, that was a typo, but I fixed, still no dice. The incoming jabber call doesn’t fire the gtalk connection. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Thursday, February 10, 2011 10:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in the other. Thanks, --Warren Selby, dCAP On Feb 10, 2011, at 5:55 PM, William Stillwell will...@stillwellsoft.com wrote: Sorry, Asterisk Build 1.6.2.7 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Thursday, February 10, 2011 6:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Gtalk/Jabber Issue OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can’t seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes ;autoprune=no autoregister=yes [jb_jabber] type=client serverhost=talk.google.com username=xx...@gmail.com/Talk secret=XXX port=5222 usetls=yes usesasl=yes ;status=Available statusmessage=Connected via Asterisk ;timeout=100 ;keepalive=yes Extensions.conf [google-in] exten = s,1,NoOp(Call from GTalk) exten = s,n,Set(CallerID(Name)=From GoogleTalk) exten = s,n,Dial(SIP/1000) jabber show connected Jabber Users and their status: User: xxx...@gmail.com/Talk - Connected Number of users: 1 CLI on incoming Call bannana*CLI JABBER: jb_jabber INCOMING: iq from=+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 to=**@gmail.com/TalkD876FAA0 id=jingle:10.218.14.137-17447266:1:03800E94 type=setses:session type=initiate id=SIP1007753261@10.218.122.83 initiator=+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 xmlns:ses=http://www.google.com/session;pho:description xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0 name=PCMU
