Re: [asterisk-users] sip.conf - sort order, does it matter
17 feb 2010 kl. 19.12 skrev Joseph: Does the sort order matter in sip.conf file? I know sort order might effect: allow=ulaw allow=alaw but does it matter where I place: insecure=invite ? The reason I'm asking is that I've loaded almost two identical (sip.conf and extension.conf) files on the same asterisk server and with one set insecure=invite is working correctly. When I load the second set of dial plan (sip.conf and extension.conf) insecure=invite is not taking effect. I get: ... username mismatch, have 4, digest has pstn- handle_request_invite: Failed to authenticate user KMIEC J You propably have a type=friend where the user part matches before you even hit the peer part, where the insecure configuration parameter matches. There is a confusion here on the From: username and the authentication username used, so there is a challenge sent. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf - sort order, does it matter
On Fri, Feb 19, 2010 at 8:54 AM, Olle E. Johansson o...@edvina.net wrote: You propably have a type=friend where the user part matches before you even hit the peer part, where the insecure configuration parameter matches. There is a confusion here on the From: username and the authentication username used, so there is a challenge sent. Is it just me, or would it be nice if a clear, understandable and unambiguous way to express codec desirata was invented? Is there a future iteration of SIP that deals with it? /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] directmedia/canreinvite/native bridging question
Hi! I'd like Asterisk to set up direct media connections for calls between clients who're both on the internet, and for calls between clients who're both on the private network, but not set up direct media connections for calls between clients on the internet and clients on the private network There is no easy solution available: Consider configuring two accounts/two lines per phone, and where possible use the local phone's diaplan to select the right one - with or without reinvite. Or next to your LAN PBX establish a remote/Internet PBX. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime extensions
Anyone know if my example of combining extensions.conf and realtime extensions is doable ?? Kind regards, Jonas. On Thu, 2010-02-18 at 20:15 +0100, jonas kellens wrote: How about something like : [mycontext] exten = 100,1,NoOp(calling 100) exten = 100,n,NoOp(going realtime) switch = Realtime/mycont...@realtime_extensions ; from here on we use realtime And then my MySQL-DB contains : `extensions_table` VALUES (1, 'mycontext', '100', n, 'Wait', '2'); `extensions_table` VALUES (2, 'mycontext', '100', n, 'NoOp', 'into RealTime'); 'extensions_table` VALUES (3, 'mycontext', '100', n, 'Playback', 'my-sound-file'); extconfig.conf has : realtime_extensions = mysql,asterisk,extensions_table Is all the above correct and possible ?? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Plan configuration in asterisk
- Chandrakant Solanki solanki.chandrak...@gmail.com wrote:
On Fri, Feb 19, 2010 at 12:21 PM, Gopalakrishnaiyer Venugopal-Q16770
venui...@motorola.com wrote:
Hi experts,
The extensions.conf has the dial plan set as
exten == _988XXX,1,Dial(DAHDI/g1/${EXTEN},20)
I want to modify this so that i can dial numbers with more than 10 digits for
example like accessing an IVR menu.
Warm Regards
Venugopal G
*
exten == _988XXX.,1,Dial(DAHDI/g1/${EXTEN},20)
--
Thanks, Phil
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Re: [asterisk-users] OpenVPN/SNOM 820: a review.
- Ken D'Ambrosio k...@jots.org wrote: Hey, all. Got an SNOM 820 in the other day to kick the tires. As with many phones, provisioning it was a bit of a PITA. The biggest problem, as far as I could tell, was that their firmware just doesn't seem that stable, and is sometimes hard to get to. - I managed to corrupt the firmware twice; fortunately, instead of bricking the phone, there's a fairly easy-to-use rescue mode. - Google was *not* your friend to find the URL to current firmware (for non-beta, it's http://wiki.snom.com/Firmware ; for beta, it's http://wiki.snom.com/Firmware/V8/Beta ) - There's a (non-standard) VPN release of firmware that has to be installed to get OpenVPN going. - Also got WLAN going; note that, apparently (and to my surprise), it appears that WPA keys are case-sensitive, and the phones default to uppercase. Beware. Also, you have to buy a ~$40 USB stick to get it going, but that sounds more awkward than it is: the phone has a nicely-recessed cavity on the bottom where it plugs in. Next, if you aren't familiar with OpenVPN, I *do not* recommend having the phone as your first client. Set up a Linux or Windows client, first, to get the hang of it. Then move on to the phone. For example, one of my firmware corruptions occurred when I named a file client.conf (.conf being the usual Linux-based OpenVPN configuration file extension), instead of client.cnf. Had to reflash. Bottom line: the phone actually works quite nicely. Provisioning for a one-off is a pain, but SNOM seems to have the hooks in place to make larger rollouts quite easy. OpenVPN works like a champ, but should be handled with care for those who don't have experience with it. The speakerphone quality is quite nice, and there are lots of nifty features the SNOM offers that I haven't seen on other phones -- for example, netcat is used for debugging OpenVPN, and a SIP log is truly nifty. One-line summary: recommended, but be prepared to spend some time getting the first one going if some of the more esoteric features (VPN, WLAN) are used. -Ken We are testing 370/870s at the moment as we have a strong requirement for OpenVPN support. We are still trying to get them to work! It would appear on the face of it that the phones use OpenVPN V1 and not V2 which is not to good. Secondly you have to create a tar ball with the configuration in side it which has to include the key. Hmmm, how would you get that to remote clients ? Put on a public webserver; not so good me thinks. Now I could be completely wrong on these things so would be very grateful for your input. -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf - sort order, does it matter
19 feb 2010 kl. 10.22 skrev Randy R: On Fri, Feb 19, 2010 at 8:54 AM, Olle E. Johansson o...@edvina.net wrote: You propably have a type=friend where the user part matches before you even hit the peer part, where the insecure configuration parameter matches. There is a confusion here on the From: username and the authentication username used, so there is a challenge sent. Is it just me, or would it be nice if a clear, understandable and unambiguous way to express codec desirata was invented? Is there a future iteration of SIP that deals with it? It's not only SIP, it's the whole Asterisk codec negotiation framework that needs a serious overhaul: Please read: http://www.voip-forum.com/asterisk/2010-02/asterisk-18-lts-wishlist-1-media-negiotiation-framework/ Interestingly enough, this blog post (and the same message on asterisk-dev) has got NO feedback, even though this has been a hot topic for years. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN/SNOM 820: a review.
- --[ UxBoD ]-- ux...@splatnix.net wrote: - Ken D'Ambrosio k...@jots.org wrote: Hey, all. Got an SNOM 820 in the other day to kick the tires. As with many phones, provisioning it was a bit of a PITA. The biggest problem, as far as I could tell, was that their firmware just doesn't seem that stable, and is sometimes hard to get to. - I managed to corrupt the firmware twice; fortunately, instead of bricking the phone, there's a fairly easy-to-use rescue mode. - Google was *not* your friend to find the URL to current firmware (for non-beta, it's http://wiki.snom.com/Firmware ; for beta, it's http://wiki.snom.com/Firmware/V8/Beta ) - There's a (non-standard) VPN release of firmware that has to be installed to get OpenVPN going. - Also got WLAN going; note that, apparently (and to my surprise), it appears that WPA keys are case-sensitive, and the phones default to uppercase. Beware. Also, you have to buy a ~$40 USB stick to get it going, but that sounds more awkward than it is: the phone has a nicely-recessed cavity on the bottom where it plugs in. Next, if you aren't familiar with OpenVPN, I *do not* recommend having the phone as your first client. Set up a Linux or Windows client, first, to get the hang of it. Then move on to the phone. For example, one of my firmware corruptions occurred when I named a file client.conf (.conf being the usual Linux-based OpenVPN configuration file extension), instead of client.cnf. Had to reflash. Bottom line: the phone actually works quite nicely. Provisioning for a one-off is a pain, but SNOM seems to have the hooks in place to make larger rollouts quite easy. OpenVPN works like a champ, but should be handled with care for those who don't have experience with it. The speakerphone quality is quite nice, and there are lots of nifty features the SNOM offers that I haven't seen on other phones -- for example, netcat is used for debugging OpenVPN, and a SIP log is truly nifty. One-line summary: recommended, but be prepared to spend some time getting the first one going if some of the more esoteric features (VPN, WLAN) are used. -Ken We are testing 370/870s at the moment as we have a strong requirement for OpenVPN support. We are still trying to get them to work! It would appear on the face of it that the phones use OpenVPN V1 and not V2 which is not to good. Secondly you have to create a tar ball with the configuration in side it which has to include the key. Hmmm, how would you get that to remote clients ? Put on a public webserver; not so good me thinks. Now I could be completely wrong on these things so would be very grateful for your input. Actually I do not think the second comment would be a issue; as like anything it comes down to how you secure your delivery. -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN/SNOM 820: a review.
Ken D'Ambrosio wrote: Hey, all. Got an SNOM 820 in the other day to kick the tires. As with many phones, provisioning it was a bit of a PITA. The biggest problem, as far as I could tell, was that their firmware just doesn't seem that stable, and is sometimes hard to get to. - I managed to corrupt the firmware twice; fortunately, instead of bricking the phone, there's a fairly easy-to-use rescue mode. - Google was *not* your friend to find the URL to current firmware (for non-beta, it's http://wiki.snom.com/Firmware ; for beta, it's http://wiki.snom.com/Firmware/V8/Beta ) - There's a (non-standard) VPN release of firmware that has to be installed to get OpenVPN going. - Also got WLAN going; note that, apparently (and to my surprise), it appears that WPA keys are case-sensitive, and the phones default to uppercase. Beware. Also, you have to buy a ~$40 USB stick to get it going, but that sounds more awkward than it is: the phone has a nicely-recessed cavity on the bottom where it plugs in. Next, if you aren't familiar with OpenVPN, I *do not* recommend having the phone as your first client. Set up a Linux or Windows client, first, to get the hang of it. Then move on to the phone. For example, one of my firmware corruptions occurred when I named a file client.conf (.conf being the usual Linux-based OpenVPN configuration file extension), instead of client.cnf. Had to reflash. Bottom line: the phone actually works quite nicely. Provisioning for a one-off is a pain, but SNOM seems to have the hooks in place to make larger rollouts quite easy. OpenVPN works like a champ, but should be handled with care for those who don't have experience with it. The speakerphone quality is quite nice, and there are lots of nifty features the SNOM offers that I haven't seen on other phones -- for example, netcat is used for debugging OpenVPN, and a SIP log is truly nifty. One-line summary: recommended, but be prepared to spend some time getting the first one going if some of the more esoteric features (VPN, WLAN) are used. -Ken What firmware version were you using? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN/SNOM 820: a review.
- Ken D'Ambrosio k...@jots.org wrote: Hey, all. Got an SNOM 820 in the other day to kick the tires. As with many phones, provisioning it was a bit of a PITA. The biggest problem, as far as I could tell, was that their firmware just doesn't seem that stable, and is sometimes hard to get to. - I managed to corrupt the firmware twice; fortunately, instead of bricking the phone, there's a fairly easy-to-use rescue mode. - Google was *not* your friend to find the URL to current firmware (for non-beta, it's http://wiki.snom.com/Firmware ; for beta, it's http://wiki.snom.com/Firmware/V8/Beta ) - There's a (non-standard) VPN release of firmware that has to be installed to get OpenVPN going. - Also got WLAN going; note that, apparently (and to my surprise), it appears that WPA keys are case-sensitive, and the phones default to uppercase. Beware. Also, you have to buy a ~$40 USB stick to get it going, but that sounds more awkward than it is: the phone has a nicely-recessed cavity on the bottom where it plugs in. Next, if you aren't familiar with OpenVPN, I *do not* recommend having the phone as your first client. Set up a Linux or Windows client, first, to get the hang of it. Then move on to the phone. For example, one of my firmware corruptions occurred when I named a file client.conf (.conf being the usual Linux-based OpenVPN configuration file extension), instead of client.cnf. Had to reflash. Bottom line: the phone actually works quite nicely. Provisioning for a one-off is a pain, but SNOM seems to have the hooks in place to make larger rollouts quite easy. OpenVPN works like a champ, but should be handled with care for those who don't have experience with it. The speakerphone quality is quite nice, and there are lots of nifty features the SNOM offers that I haven't seen on other phones -- for example, netcat is used for debugging OpenVPN, and a SIP log is truly nifty. One-line summary: recommended, but be prepared to spend some time getting the first one going if some of the more esoteric features (VPN, WLAN) are used. -Ken Would be nice if the VPN support could be back ported to the 360s. -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registering of Asterisk against a SIP provider
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday, Feb 18, 2010 at 05:36:41 -0300, Administrator TOOTAI wrote: Hi Hi, Daniel. Daniel Bareiro a écrit : [...] Hours ago the IP changed and the domain was updated satisfactorily, but in spite of this I was obtaining the registering failures that I mentioned above. After to restart Asterisk (1.4.24.1), I no longer had this problem of registering. But there would be some way to solve this problem? [...] It's an old story. Asterisk check DNS when it start that's why it's ok after you have it restarted. When I was running Asterisk using dynamic addresses, I made following: - modify sip.conf to include a file placed where ever you want, contents being externalip/externalhosts and all others info needed related to external IP - restarted myself ADSL line with a cron script each night - this script extract/found the new IP using the method you prefer (eg ping your dyndns host until response and than you have your new IP and insert the IP in the file you include in sip.conf - this script restart asterisk and voila :-) Was working like a charm. As I said to Warren, according to the tests that I was doing, apparently this can be solved with both externip and externhost,restarting Asterisk in either cases. In the case of externhost we would be saving ourselves to have to modify the IP in sip.conf every time, but even so we would have to verify if the IP has changed for restarting Asterisk. I thought that perhaps this could be solved without restarting Asterisk. Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkt+kkYACgkQZpa/GxTmHTcpxwCfbwAbaYEzEv6rBqZIWQs5kLER STkAn00FEXGzD+berHCZYe20HLBnXZQU =xzIk -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Volume of Playback() application
Hello, Anyone know how I can intesify volume of an application playback()? Thank you very much. ye Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Volume of Playback() application
AFAIK, playback/background has no gain adjustability. Two possible work-arounds would be to adjust gain on the line/extension or to use sox to create a louder version of the file you want to playback sox v +2 vm-goodbye.gsm vm-goodbye2.gsm _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Renato bianchini Sent: Friday, February 19, 2010 7:57 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Volume of Playback() application Hello, Anyone know how I can intesify volume of an application playback()? Thank you very much. ye _ Veja quais são os assuntos do momento no Yahoo! + Buscados: Top http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ 10 - Celebridades http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ celebridades/ - Música http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ m%C3%BAsica/ - Esportes http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/ esportes/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI + device status (patch 0016732) + remote control
Hi all, we are looking for a solution which transfers device status changes (events) like busy, picked up, detected answering machine/fax and no valid number to another server which takes action according to these events. I already found the patch mentioned in the subject of this message. Which sounds quite like it does, what we are looking for: https://issues.asterisk.org/view.php?id=16732 If you think that we are right using AMI for that, please tell me. If you think another approach is more suitable, say so. I really appreciate any hints on the stated issue. Thanks a lot in advance. Cheers, Marcus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how asterisk knows which context forward the call to?
I think you are correct, thank you for pointing it out. I just switch entries in sip.Cong put [pstn-9998] first' and [pstn-] second and the second entry was selected :-( (so you are right on). Audiocodes gateway, has two FXO ports, I was convinced that entry is selected based on registration context in [square-bracket] of sip.conf but it doesn't appear to be the case; the last registered entry is selected as default. Is it a limitation how SIP works or asterisk limitation? Is it possible to split registration into two different sip.conf files (sip1.conf and sip2.conf)? -- Joseph On 02/19/10 09:05, Ioan Indreias wrote: I hope I'm not wrong but I think the problem is related to the fact that on incoming calls Asterisk find the peers based on their IP and not on their IP+PORT. Thus, if you have several extensions on the same devices (= one single IP with different SIP ports), the last entry into your sip.conf file is taken into consideration = all calls are sent to the context of that last extension. You could check this if you configure a higher verbose/debug level (like more than 10) and check into the Asterisk logs the information displayed by chan_sip.c HTH, Ioan Indreias www.modulo.ro ### extract from: http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels ### Incoming SIP Connections === When Asterisk receives an incoming SIP call, the SIP Channel Module + first tries to find a [user] section matching the caller name (From: username), + then tries to find a [peer] section matching the caller's IP address. + If no matching user or peer is found, the call is sent to the context defined in the [general] section of sip.conf. See: Asterisk SIP user vs peer ### On Fri, Feb 19, 2010 at 7:48 AM, Joseph syscon...@gmail.com wrote: Yes, it should but it doesn't. And the gurus at Audiocodes support can not explain why? -- Joseph On 02/18/10 19:27, C F wrote: It should use the context of the device On Wed, Feb 17, 2010 at 8:40 PM, Joseph syscon...@gmail.com wrote: Is there any asterisk guru who can explain me how how asterisk knows which context forward the call to? -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Plan configuration in asterisk
19 feb 2010 kl. 11.47 skrev --[ UxBoD ]--:
exten == _988XXX.,1,Dial(DAHDI/g1/${EXTEN},20)
UxBoD - you really have to read the security advisory before sending out such
examples on the mailing list. Please go to http://www.asterisk.org now.
Have a nice weekend!
Thanks,
/O
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Re: [asterisk-users] [SPAM] - Re: Asterisk t38modem Fax gateway evaluation - Email found in subject
Hi, Many thanks Steve and Philipp for your input. I am compiling Asterisk 1.6 svn version right now and will try to integrate the T.38 Gateway patches mentioned at https://issues.asterisk.org/view.php?id=13405 (I have some Linux coding experience, but unfortunately my telecommunication knowledge is severly lacking). The ISDN PRI - Analogue Converter solution mentioned by Philipp is impractical for my situation as the hub of the 2-wire POTS lines is not in the same Building as the PRI. I will let you her if my research bears some practical results, as this seems to be something not very well established/documented. Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mISDN (HFC-S) and TDM400P - isac xdu no tx_busy
I had Asterisk 1.6.2.2 running fine with a mISDN using a HFC-S based card. I installed my TDM400P into the PC, it's really slow to boot now, when it finally does I gets stuck in a loop of reporting isac xdu no tx_busy. Anyone able to assist? Thanks in advance! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom VVX1500 video working yet?
On 18 February 2010 00:14, Michael Graves mgra...@mstvp.com wrote: On Wed, 17 Feb 2010 17:12:01 +, Steve Davies wrote: On 17 February 2010 16:56, asterisk aster...@nbsvoice.com wrote: Can anyone tell if asterisk and Polycom VVX1500 work with video yet? If so what version? Is there a patch? Thank you! Doug According to my experimentation, Polycom VVX1500 phones work with all versions of Asterisk as far back as 1.2.30, and possibly older. This is with the earliest of the Polycom firmware to support this device. The problem (cause not known) is that Polycom VVX1500 phones only talk to other Polycom VVX1500 phones, making them essentially useless. Perhaps Polycom have fixed this in newer firmware versions - I've not seen such a fix in their changelogs. This is simply untrue. However, the VVX-1500 is fussy about the video that it receives. It supports only CIF and SIF resolutions. Period. Send it anything else and you'll likely crash it. [snip] I just upgraded to the new bootblock and 3.2.2 firmware, and these phones will now talk video to other devices. Nothing in the changelogs indicates why, but there is a definite jump up from the previous release of this phone. So, I duly stand corrected. OTOH, once upgraded, DHCP seems to be broken on these phones where static IP configuration is working perfectly :( Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] transcoding with TC400P
Hello, I have transcoding card TC400P installed in server running Debian with Asterisk 1.4.23. Everything seams to be fine and after I boot up server I see in dmesg: 7.590966] Zapata Telephony Interface Registered on major 196 [7.590966] Zaptel Version: 1.4.12.1 [7.590966] Zaptel Echo Canceller: MG2 [7.610963] zttranscode: Loaded. [7.618969] wctc4xxp: tc400b0: Attached to device at :02:07.0. [7.618969] firmware: requesting zaptel-fw-tc400m.bin [8.86] ACPI: PCI Interrupt :02:07.0[A] - GSI 19 (level, low) - IRQ 19 [ 11.400359] wctc4xxp: tc400b0: (G.729a / G.723.1) Transcoder support LOADED (firm ver = 6.12) [ 11.400359] wctc4xxp: tc400b0: Installed a Wildcard TC: Wildcard TC400P+TC400M [ 11.400359] zttranscode: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) [ 11.400359] zttranscode: Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=0101, dsts=000c) [ 11.572355] ACPI: PCI Interrupt :00:1f.5[B] - GSI 17 (level, low) - IRQ 17 in asterisk cli: Connected to Asterisk 1.4.23.1 currently running on katerin (pid = 3168) Verbosity was 3 and is now 5 katerin*CLI transcoder show 0/0 encoders/decoders of 92 channels are in use. I am trying to do 2 experiments: 1) asterisk1 ---g729-- asterisk2 (with transcoding card) Playfile in ulaw format 2) asterisk2 ---ulaw-- asterisk2 (with transcoding card) Playfile in g729 format In the first case I get all calls proceeding and in asteris2 cli Connected to Asterisk 1.4.23.1 currently running on katerin (pid = 3168) Verbosity was 3 and is now 5 katerin*CLI transcoder show 90/0 encoders/decoders of 92 channels are in use. But it all does not work for the second case, in asterisk2 cli I get messages like [Feb 19 15:18:32] ERROR[3121]: codec_dahdi.c:244 zap_translate: Unable to attach to transcoder: Input/output error [Feb 19 15:18:32] WARNING[3121]: translate.c:294 ast_translator_build_path: Failed to build translator step from 8 to 2 [Feb 19 15:18:32] WARNING[3121]: chan_sip.c:3751 sip_write: Asked to transmit frame type 256, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x100 (g729)(256) Btw call does go through but transcoding is done by processor not by TC400P. Did anyone encounter such problem? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] transcoding with TC400P
- Katerina Borin katerin.bo...@gmail.com escreveu: Hello, I have transcoding card TC400P installed in server running Debian with Asterisk 1.4.23. Everything seams to be fine and after I boot up server I see in dmesg: 7.590966] Zapata Telephony Interface Registered on major 196 [7.590966] Zaptel Version: 1.4.12.1 [7.590966] Zaptel Echo Canceller: MG2 [7.610963] zttranscode: Loaded. [7.618969] wctc4xxp: tc400b0: Attached to device at :02:07.0. [7.618969] firmware: requesting zaptel-fw-tc400m.bin [8.86] ACPI: PCI Interrupt :02:07.0[A] - GSI 19 (level, low) - IRQ 19 [ 11.400359] wctc4xxp: tc400b0: (G.729a / G.723.1) Transcoder support LOADED (firm ver = 6.12) [ 11.400359] wctc4xxp: tc400b0: Installed a Wildcard TC: Wildcard TC400P+TC400M [ 11.400359] zttranscode: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) [ 11.400359] zttranscode: Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=0101, dsts=000c) [ 11.572355] ACPI: PCI Interrupt :00:1f.5[B] - GSI 17 (level, low) - IRQ 17 in asterisk cli: Connected to Asterisk 1.4.23.1 currently running on katerin (pid = 3168) Verbosity was 3 and is now 5 katerin*CLI transcoder show 0/0 encoders/decoders of 92 channels are in use. I am trying to do 2 experiments: 1) asterisk1 ---g729-- asterisk2 (with transcoding card) Playfile in ulaw format 2) asterisk2 ---ulaw-- asterisk2 (with transcoding card) Playfile in g729 format In the first case I get all calls proceeding and in asteris2 cli Connected to Asterisk 1.4.23.1 currently running on katerin (pid = 3168) Verbosity was 3 and is now 5 katerin*CLI transcoder show 90/0 encoders/decoders of 92 channels are in use. But it all does not work for the second case, in asterisk2 cli I get messages like [Feb 19 15:18:32] ERROR[3121]: codec_dahdi.c:244 zap_translate: Unable to attach to transcoder: Input/output error [Feb 19 15:18:32] WARNING[3121]: translate.c:294 ast_translator_build_path: Failed to build translator step from 8 to 2 [Feb 19 15:18:32] WARNING[3121]: chan_sip.c:3751 sip_write: Asked to transmit frame type 256, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x100 (g729)(256) Btw call does go through but transcoding is done by processor not by TC400P. Did anyone encounter such problem? I never used a TC400 card so I don't have much knowledge on that, but you really shouldn't be using Zaptel anymore. Upgrade to DAHDI, that might solve your problem. Atenciosamente, Vinícius Fontes Gerente de Segurança da Informação Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] splitting sip.conf to two files
Is it possible to split sip.conf into two files (sip1.conf sip2.conf)? I have an Audiocodes gateway with two FXO ports, and (according to info I received, and it appears to be correct) Asterisk find the peers based on their IP and not on their IP+PORT. Thus, Audiocodes with two FXO ports registered on the same devices (= one single IP with different SIP ports), the last entry into my sip.conf file is taken into consideration = all calls are sent to the context of that last extension. So I can only use one context for incoming calls. If I split the sip.conf into two files will it make any difference. Is it a limitation/bug in Asterisk or sip.conf? -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] splitting sip.conf to two files
I think sip.conf will allow the inclusion of a second (or greater) sip2.conf file. This might only apply to extensions.conf, but I'm betting all .conf files are processed with the same parser. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Friday, February 19, 2010 10:22 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] splitting sip.conf to two files Is it possible to split sip.conf into two files (sip1.conf sip2.conf)? I have an Audiocodes gateway with two FXO ports, and (according to info I received, and it appears to be correct) Asterisk find the peers based on their IP and not on their IP+PORT. Thus, Audiocodes with two FXO ports registered on the same devices (= one single IP with different SIP ports), the last entry into my sip.conf file is taken into consideration = all calls are sent to the context of that last extension. So I can only use one context for incoming calls. If I split the sip.conf into two files will it make any difference. Is it a limitation/bug in Asterisk or sip.conf? -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Virtual machine timing (KVM)
To get MeetMe working properly, I know some sort of timing device provided by the zaptel package is required (even if it means the zt_dummy). But, on a virtual machine I know that the Linux timing won't work as expected. Is it possible to then dedicate a physical device like a USB port or something to the virtual machine to use for the timing interrupts? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] splitting sip.conf to two files
I'm suspecting you might be correct; so it will not make much difference. -- Joseph On 02/19/10 10:29, Danny Nicholas wrote: I think sip.conf will allow the inclusion of a second (or greater) sip2.conf file. This might only apply to extensions.conf, but I'm betting all .conf files are processed with the same parser. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Friday, February 19, 2010 10:22 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] splitting sip.conf to two files Is it possible to split sip.conf into two files (sip1.conf sip2.conf)? I have an Audiocodes gateway with two FXO ports, and (according to info I received, and it appears to be correct) Asterisk find the peers based on their IP and not on their IP+PORT. Thus, Audiocodes with two FXO ports registered on the same devices (= one single IP with different SIP ports), the last entry into my sip.conf file is taken into consideration = all calls are sent to the context of that last extension. So I can only use one context for incoming calls. If I split the sip.conf into two files will it make any difference. Is it a limitation/bug in Asterisk or sip.conf? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN/SNOM 820: a review.
--[ UxBoD ]-- wrote: Would be nice if the VPN support could be back ported to the 360s. Never going to happen, there isn't enough flash memory to store the code. The Snom370 has had OpenVPN support for quite a while though. cheers, Paul. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual machine timing (KVM)
On Fri, Feb 19, 2010 at 11:42 AM, Mike A. Leonetti mleone...@evolutionce.com wrote: To get MeetMe working properly, I know some sort of timing device provided by the zaptel package is required (even if it means the zt_dummy). But, on a virtual machine I know that the Linux timing won't work as expected. Is it possible to then dedicate a physical device like a USB port or something to the virtual machine to use for the timing interrupts? You could always use ConfBridge(), starting in 1.6.2.*, which does not require DAHDI/Zaptel, and therefore doesn't require a timer. Let me be the first to tell you that using a virt for a conferencing solution, especially if you want people to actually use it, sounds like a 'Bad Idea'. You could oversubscribe the resources so you don't starve the virt, but we already have a name or that. It's called not using a virt in the first place. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual machine timing (KVM)
David Backeberg wrote: You could always use ConfBridge(), starting in 1.6.2.*, which does not require DAHDI/Zaptel, and therefore doesn't require a timer. It *does* require a timer (all conferencing requires a timer), but it does not require a DAHDI/Zaptel timer, there are other options available. MeetMe not only requires a timer, the mixing itself is done in DAHDI/Zaptel, whereas ConfBridge does the mixing in the application. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] string length in dialplan
I am trying to find out how I can tell the length of a string actually CALLERID(num) in the dialplan. How is that done? If need to test the length of the CALLERID(num) if its less the 10 digits I need to set it to a known value or insert 0's at the beginning until it is 10 digits in length. My PRI provider needs it set to 10 digits always. Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual machine timing (KVM)
To get MeetMe working properly, I know some sort of timing device provided by the zaptel package is required (even if it means the zt_dummy). But, on a virtual machine I know that the Linux timing won't work as expected. Is it possible to then dedicate a physical device like a USB port or something to the virtual machine to use for the timing interrupts? The 2.2.1 version of DAHDI using DAHDI dummy seems to be working adequately in a Xen environment on CentOS for me, although I haven't been using MeetMe. Have you run into issues with it specifically? Which version of DAHDI are you using? If there are some issues that you have found I would like to know... Thanks, Sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] string length in dialplan
Jerry Geis escribió: I am trying to find out how I can tell the length of a string actually CALLERID(num) in the dialplan. How is that done? If need to test the length of the CALLERID(num) if its less the 10 digits I need to set it to a known value or insert 0's at the beginning until it is 10 digits in length. My PRI provider needs it set to 10 digits always. Thanks, Jerry *CLI core show function LEN -= Info about function 'LEN' =- [Syntax] LEN(string) [Synopsis] Returns the length of the argument given [Description] Not available Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] string length in dialplan
I am trying to find out how I can tell the length of a string actually CALLERID(num) in the dialplan. How is that done? CLI: show function LEN -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk sudden restart - 1.4.18.1 - Resolved after upgrade to 1.4.22.1
Hi Leif, Thanks for the information. I checked the /tmp/ folder and there was core files and I tried to back trace it but it was not showing the cause of that crash, but anyhow, I upgraded our Asterisk system to 1.4.22.1 and from past few days its going on fine. I have also researched and found that version 1.4.17/18.1 had the issue of channel stuck up as well as random asterisk crashes. Regards Sandesh On Thu, Feb 11, 2010 at 6:29 AM, Leif Madsen leif.mad...@asteriskdocs.orgwrote: das sandesh wrote: Hi, Asterisk got stopped this morning after 20 minutes and phones went to 'No Service' and then got started automatically after 20 min, as I could see in the full log that asterisk got started at so and so time: [Feb 10 08:29:31] VERBOSE[31013] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log [Feb 10 08:29:31] VERBOSE[31013] logger.c: Asterisk Dynamic Loader Starting: But I am trying to find why did it stopped (and there was no record of asterisk stopped?) and then get restarted.In the log I could also see : [Feb 10 08:02:05] VERBOSE[7027] logger.c: -- Incoming call: Got SIP response 500 CSeq Number Out of order back from 192.168.10.16 [Feb 10 08:02:05] VERBOSE[7027] logger.c: -- Incoming call: Got SIP response 500 CSeq Number Out of order back from 192.168.10.16 [Feb 10 08:02:05] VERBOSE[7027] logger.c: -- Incoming call: Got SIP response 500 CSeq Number Out of order back from 192.168.10.16 [Feb 10 08:02:08] VERBOSE[7004] logger.c: -- Remote UNIX connection [Feb 10 08:02:08] VERBOSE[28272] logger.c: -- Remote UNIX connection disconnected During this period (from 8:02 till 8:29) all the phones went to 'No Service'I checked all the logs and could not find any reason why it was down or any log that shows asterisk was down at that point..any ideas are appreciated... Check /tmp/ as there may be a core.# file there which you could generate a backtrace from to determine the issue (if you're able to understand what is outputs :)) At the least, if such a core file exists, then you could line it up with the time to see if that was indeed the case. If there is a core file, it means something caused Asterisk to crash. Asterisk version: 1.4.18.1 This version is quite old, and if the issue was a crash that brought the system down momentarily, it is possible this issue may already be resolved. Leif! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] splitting sip.conf to two files
On Fri, Feb 19, 2010 at 09:21:46AM -0700, Joseph wrote: Is it possible to split sip.conf into two files (sip1.conf sip2.conf)? I have an Audiocodes gateway with two FXO ports, and (according to info I received, and it appears to be correct) Asterisk find the peers based on their IP and not on their IP+PORT. Thus, Audiocodes with two FXO ports registered on the same devices (= one single IP with different SIP ports), the last entry into my sip.conf file is taken into consideration = all calls are sent to the context of that last extension. So I can only use one context for incoming calls. If I split the sip.conf into two files will it make any difference. Is it a limitation/bug in Asterisk or sip.conf? I assume you use '#include to separate sip.conf to two files. #include is a verbatim inclusion, and thus for all prictical purposes it is the same as if everything were in a single file. Configuration [sections] cannot be repeated in the Asterisk configuration files. If you want to add later on anything to [foo], you can't just add a second [foo] . Rather, you should add: [foo](+) This will add the content of that section after the content of the existing section [foo]. See http://svn.digium.com/svn/asterisk/trunk/doc/tex/configuration.tex (Any better direct link?) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error redirecting an incoming call of a SIP provider to a local extension
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I am trying to redirect to a local extension the incoming calls that I
receive to an account which I have in iptel.org, but when receiving I'm
obtaining this error:
alderamin*CLI
-- Executing [...@from-internal:1] Dial(SIP/danib-089f8820,
SIP/300|30|tTrm) in new stack
[Feb 19 19:22:50] WARNING[19254]: app_dial.c:1237 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
[Feb 19 19:23:00] WARNING[19254]: pbx.c:2529 __ast_pbx_run: Timeout, but
no rule 't' in context 'from-internal'
It is probable that this can be due to a problem of interaction between
contexts? I copy the content of extensions.conf and sip.conf to see if
it can help to find the problem:
-
extensions.conf:
; DGB - 20091114
[general]
autofallthrough=no
[macro-dial]
exten = s,1,Dial(${ARG1},15)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${macro_ext...@voicemail,u)
exten = s-NOANSWER,n,Hangup
exten = s-BUSY,1,Voicemail(${macro_ext...@voicemail,b)
exten = s-BUSY,n,Hangup
exten = s-CHANUNAVAIL,1,Playback(pbx-invalid)
[from-internal]
; Llamadas a extensiones SIP
exten = _2xx,1,Macro(dial,SIP/${EXTEN})
exten = _2xx,n,Hangup
exten = 300,1,Dial(SIP/300,30,tTrm)
; Extension analogica
exten = 402,1,Macro(dial,DAHDI/2)
exten = 402,n,Hangup
; Directorio de extensiones
exten = *400,1,Directory(voicemail,from-internal)
; Musica en espera
exten = *300,1,Answer
exten = *300,n,SetMusicOnHold(default)
exten = *300,n,WaitMusicOnHold(2000)
exten = *300,n,Hangup
; Prueba de Eco
exten = *200,1,Answer
exten = *200,n,Playback(demo-echotest)
exten = *200,n,Echo
exten = *200,n,Playback(demo-echodone)
exten = *200,n,Hangup
; Acceso a voicemail
exten = *100,1,Answer
exten = *100,n,Wait(1)
exten = *100,n,VoiceMailMain(${CALLERID(num)}...@voicemail)
exten = *100,n,Hangup
; Llamadas salientes
exten = _9.,1,Dial(DAHDI/1/${EXTEN:1})
exten = _9.,n,Hangup
; Call a number at iptel.org
exten = _0.,1,Dial(SIP/iptel/${EXTEN:1},20,r))
exten = _0.,n,Hangup
[from-pstn]
; incoming calls from FXO port are directed to this context
exten = s,1,Answer()
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=15)
exten = s,n,Background(contestador1)
exten = i,1,Goto(from-pstn,s,1)
exten = t,1,Playback(locomunicoconelinterno1)
exten = t,n,Dial(SIP/200,25)
exten = t,n,VoiceMail(2...@voicemail,20)
exten = t,n,Hangup()
include = from-internal
-
sip.conf:
[general]
[...]
; register with iptel.org
register = danib:mlrzv...@iptel.org/300
[...]
; Outgoing to iptel.org
[iptel]
type=friend
username=danib
secret=myspasswd
host=iptel.org
canreinvite=no
qualify=300
insecure=port,invite ; required for incoming ekiga.net calls
context = from-internal
-
Thanks in advance for your replies.
Regards,
Daniel
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Re: [asterisk-users] How to run a remote PHP script and still have access to audio stream?
On Wed, Feb 10, 2010 at 2:13 PM, Leo Burd l...@media.mit.edu wrote:
Hello David,
Thanks so much for your message!
Please check my comments inline below...
David Backeberg wrote:
On Sun, Feb 7, 2010 at 9:54 PM, Leo Burd l...@media.mit.edu wrote:
Hello there,
I'm trying to figure out how to run a PHP script on a remote machine and
still have access to the audio stream associated with the call.
Ideally, I'd love to play/record audio files directly from/to the remote
server without having to copy them back and forth to the Asterisk
server. What is the best way to do this?
1) recordings, with a side order of distributing those to another machine
2) remote shell scripting
What would be the asterisk way of recording part of the call from a
remote server? I'm not sure I can do that (the remote connection) with
EAGI, can I?
The 'asterisk way' of recording part of a call needs to be done on the
asterisk system where the call is taking place. Or, if the call is
actually between two asterisk systems, the call can be recorded
directly on one or the other or both asterisk systems. The asterisk
dialplan feature is called Monitor() or MixMonitor(), and you can
refer to the documentation for the differences.
The AGI and remote connection comes in when the recording (call)
completes, and in the h (hangup) context for this dialplan context,
you would do the remote file copy so your call would now be copied off
somewhere else.
Do you know of any examples that use ssh from inside Asterisk calls?
Sure, here's an example from one of my dialplans.
exten = s,1,Answer
exten = s,n,Set(CDR(userfield)=faxsample)
exten = s,n,Set(LOCALSTATIONID=FaxSample)
exten = s,n,Set(LOCALPATH=/var/spool/fax/recvq/${LOCALSTATIONID}/)
exten = s,n,Set(MYDATE=${STRFTIME(${EPOCH},,%C%y-%m-%d-%H%M)})
exten = s,n,Set(MYFILENAME=${LOCALSTATIONID}-${MYDATE}-${CDR(uniqueid)})
exten = s,n,Set(MYFULLPATH=${LOCALPATH}${MYFILENAME})
exten = s,n,Set(KEYFULLPATH=/var/spool/fax/ssh_key_for_remote_copy)
exten = s,n,Set(SCPUSER=filecopyuser)
exten = s,n,Set(FILESERVER=fileserver.domain.com)
exten = s,n,Set(REMOTEPATH=/path/to/where/it/should/go/${LOCALSTATIONID})
exten = s,n,Set(RECORDING=${LOCALPATH}recording/${MYFILENAME}.gsm)
exten = s,n,MixMonitor(${RECORDING})
exten = s,n,Playback(silence/1)
exten = s,n,ReceiveFax(${MYFULLPATH}.tif)
; log what happened with the fax transmission
exten = h,1,System(/bin/echo ${MYFULLPATH},${CALLERID(num)},${CALLERID(name)},$
{FAXSTATUS},${FAXERROR},${FAXMODE},${FAXPAGES},${REMOTESTATIONID}
${LOCALPATH}fax.log)
;
; if fax is salvageable, a tif will exist.
exten = h,n,System(test -e ${MYFULLPATH}.tif)
;exten = h,n,Verbose(System call result was ${SYSTEMSTATUS})
;
; try to turn any file that exists into a pdf
exten = h,n,ExecIf($[${SYSTEMSTATUS} =
SUCCESS]?System(/usr/bin/tiff2pdf -pletter ${MYFULLPATH}.tif -o
${MYFULLPATH}.pdf))
;
; check if pdf exists. If the fax was too incomplete to process, no
file will exist.
; If yes, send it off to H drive
exten = h,n,System(test -e ${MYFULLPATH}.pdf)
;exten = h,n,Verbose(System call result was ${SYSTEMSTATUS})
exten = h,n,ExecIf($[${SYSTEMSTATUS} = SUCCESS]?System(/usr/bin/scp
-i ${KEYFULLPATH} ${MYFULLPATH}.pdf
${scpus...@${fileserver}:${REMOTEPATH}))
; if tiff file exists from good fax, name it one thing
exten = h,n,ExecIf($[${FAXSTATUS} = SUCCESS]?System(/bin/mv
${MYFULLPATH}.tif /var/spool/fax/recvq/processed))
; if fax was bad, check if we still have tif. If so, move it out
exten = h,n,System(test -e ${MYFULLPATH}.tif)
;exten = h,n,Verbose(System call result was ${SYSTEMSTATUS})
exten = h,n,ExecIf($[${SYSTEMSTATUS} = SUCCESS]?System(/bin/mv
${MYFULLPATH}.tif
/var/spool/fax/recvq/processed/${MYFILENAME}-FAILED.tif))
exten = h,n,Hangup
How much control do the ssh processes have over the call, if any?
As you can see in that particular example, I was using scp to do a
remote file copy of the received fax. I also setup MixMonitor()
against the channel with a filename that would match the cdr of the
call in case I ever needed to go back and troubleshoot a particular
fax.
Is that comparable to Fast_AGI? Or EAGI?
Ummm, kindof. My example shows doing everything directly in asterisk
dialplan. AGI let's you use the language you prefer to do arbitrary
things with calls, using the AGI library for that language. Some
people prefer AGI, some people prefer dialplan. They both have their
strengths, and drawbacks.
Strength of dialplan is it's extremely debuggable. Strength of AGI is
that you get to leverage the syntax and libraries in a language you
already use, but when you want to debug, you have (in my opinion) less
introspection into what's going on unless you use the innate debugging
native to your language of choice. ('agi set debug on' helps).
My personal experience has been that I really prefer AGI when I need
to make lots of database calls, as that's where I much prefer Perl,
and much dislike the asterisk syntax. So I have a set of AGIs that use
the
Re: [asterisk-users] How to run a remote PHP script and still have access to audio stream?
On Wed, Feb 10, 2010 at 2:13 PM, Leo Burd l...@media.mit.edu wrote: How much control do the ssh processes have over the call, if any? It occurred to me that I might be answering this backwards. So from the perspective of server A, trying to talk to a remote system B running asterisk, server A can invoke: asterisk -rx do something on the asterisk cli and it will be done to asterisk on that system. So for example, I have built a nifty web gui that displays current call status in the system, along with a bunch of buttons. Among those buttons, is one that will hangup a call, on an appropriate channel, as corresponds to the database state I've been maintaining. And this does happen to have been done in PHP. And to do this hangup, I actually do NOT run ssh with keys, but rather I use asterisk manager. And send I use a nice PHP Asterisk manager library that somebody else wrote to manage the connection, then I send the Hangup() command on the appropriate channel, and the PHP Asterisk manager takes care of the dirty work of closing and cleaning up the connection. I chose to use PHP and asterisk manager, but I could have done the same thing with ssh keys and asterisk -rx '' approach. Hopefully that helps. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] splitting sip.conf to two files
Joseph wrote: Is it possible to split sip.conf into two files (sip1.conf sip2.conf)? I have an Audiocodes gateway with two FXO ports, and (according to info I received, and it appears to be correct) Asterisk find the peers based on their IP and not on their IP+PORT. Thus, Audiocodes with two FXO ports registered on the same devices (= one single IP with different SIP ports), the last entry into my sip.conf file is taken into consideration = all calls are sent to the context of that last extension. So I can only use one context for incoming calls. If I split the sip.conf into two files will it make any difference. there might be an include directive in sip.conf (i can't confirm) however Asterisk will see it as one big sip.conf so it will do absolutely nothing for you in this situation. what you can do is setup automatic dial to different extensions on the 2 ports on audiocodes. -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hung channel problem with 1.4.26.2
Hi, I have a case where SIP channels will not be destroyed, resulting in further calls to ChanIsAvail() to fail. The process (I believe) to replicate this is the following: - Make a call to another SIP phone that is an intercom call (Auto-Answer) - For whatever reason, the phone happens to go UNREACHABLE during this call - Phone comes back REACHABLE, but channel still exists in core show channels As an example, here's 3 stuck calls from today: r...@hades:~# asterisk -rx core show channels Channel Location State Application(Data) SIP/6296-a2298 (None) Up AppDial((Outgoing Line)) SIP/6315-a0906 *806...@ext-in Up Dial(SIP/6296|5|A(beep)) SIP/6333-a131e (None) Up AppDial((Outgoing Line)) SIP/6294-a24fc *806...@ext-in Up Dial(SIP/6333|5|A(beep)) SIP/6297-a1cb7 (None) Up AppDial((Outgoing Line)) SIP/6315-adc5d *806...@ext-in Up Dial(SIP/6297|5|A(beep)) I don't know if this has been fixed in a later 1.4.x version, though after reading some of the DTMF relaying problems with 1.4.27 and beyond, I don't think I would want to upgrade... Thanks. -- James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] splitting sip.conf to two files
On 02/19/10 18:38, Edwin Lam wrote: So I can only use one context for incoming calls. If I split the sip.conf into two files will it make any difference. there might be an include directive in sip.conf (i can't confirm) however Asterisk will see it as one big sip.conf so it will do absolutely nothing for you in this situation. what you can do is setup automatic dial to different extensions on the 2 ports on audiocodes. I already have setup automatic dialing, it does noting. But the solution might be to specify different port number in the Tel to IP routing table, and setup sip.conf entries to listen on these ports. Calls coming from Trunk Group 1 are to be sent on port 5065, and all calls coming from Trunk Group 2 will be sent on Trunk Group 5066. It will take two routing table entries to do this. -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
