Re: [Asterisk-Users] 0.2.0-RC8o (* 1.0.9) + No Caller ID
Non ho visto che c'era la nuova versione visto che la home dava come stabile la RC8o -- I havent seen new versione because junghanns's home report RC8o as stable version 2005/10/24, Massimo De Nadal [EMAIL PROTECTED]: Giovanni Miano wrote: I've 2 hfc billion and one TDM400P 1fxs/1fxo with bristuff 0.2.0-RC8o and * 1.0.9 I dont recive callerid from TDM400P fxo port but isdn hasnt problems If i try to use only TDM400P 1fxs/1fxo without bristuff.. all work ok is it bug of bristuff ? Maybe, why not try bristuff 0.2.0-RC8p ? For me works fine (tdm400p cid detection). maxx ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip provider in a box
Dear trixter Our software AstBill is now in use/beeing implemented by many smaal service providers and a few very large. It is Open Source. I love to work with you on this and if any features are missing we be happy to implement it. Are Casilla -- http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk and Drupal Consultants http://astbill.com - Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com On 10/22/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: I am tasked with evaluating ready made solutions for a voip provider.Does anyone have any recommendations for software, specifically theenvironment will be a chargable voip provider (ie broadvoice, vonage,etc).They wanted me to see what was there and write something if nothing they like exists.Thanks--Trixter http://www.0xdecafbad.com Bret McDanelUK +44 870 340 4605 Germany +49 801 777 555 3402US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.1 (GNU/Linux)iD8DBQBDWjJh+1olxlzQw5cRAiaVAJ47j+iPhoQ1bBIpHdX4L+w/3gvfpACfUcfqme9ecSPfEqNVSfqlvNMsFZc==UATX-END PGP SIGNATURE- ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X101P and UK CallerID...does it work?
Can anyone please let me know if they have got UK CallerID working using a X101P? While you're waiting for a live answer, there are several threads on this list you could search for. Try this http://www.google.com/search?hs=7Adhl=enq=asterisk+uk+calleridbtnG=Search ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SER for dummies ?
Dear Ralf We have a few large installations that are using Asterisk and SER managed by our Open Source software AstBill. It is working exelent. Basically Asterisk is handeling the PSTN and Voicemail part. The authentication in Asterisk is done using ANI/CLI. This setup is not very well documented yet so we have to work together for you to get it running. But it is a very powerfull and stable combination. Just contact me off list and I give you more info. Are Casilla -- http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk and Drupal Consultants http://astbill.com - Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X101P and UK CallerID...does it work?
On Tue, Oct 25, 2005 at 06:01:55AM +0100, Paul Duffy wrote: Hi All Can anyone please let me know if they have got UK CallerID working using a X101P? If so please can you let me know which version of asterisk, did you apply the UK CID patch, what are your settings in zapata.conf, zaptel.conf and extensions.conf to get it working? There is a lot of confusion regarding whether the X101P supports polarity reversal and I've read you can use the usehist setting but no matter what way I try to configure the settings (and there are about 3 different variants documented) I can't get it to work. Any help gratefully received. There is a patch there that works nicely with asterisk 1.0 and will require a bit of work to apply to 1.2 . Apart from that, what you need in zapata.conf: cidsignalling=v23 ; this only works with the X100P callerID patch: ; the TDM400P can identify this by hardware and thus needs here ; cidstart=polarity cidstart=history -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip provider in a box
On Tue, 2005-10-25 at 08:21 +0100, Are wrote: Dear trixter Our software AstBill is now in use/beeing implemented by many smaal service providers and a few very large. It is Open Source. I love to work with you on this and if any features are missing we be happy to implement it. I didnt initially want to use it because of the mysql 5.x requirement, however since I originally posted that mysql 5.x became 'stable' and I might consider it. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SER for dummies ?
On Tue, 2005-10-25 at 08:27 +0100, Are wrote: The authentication in Asterisk is done using ANI/CLI. Same way as broadvoice, wonder if using that setup if I set my caller id to someone else will it cause the INVITE that broadvoice does (broadvoice will invite the person registered as that account if you try to make a call on their CID, asterisk ignores that invite, I am not so sure if all devices will) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone using Java SIP communicator with Asterisk ?
Hi, I gave it a quick try (audio only): - set Public SIPaddress, SIP registrar, SIP-proxy etc. to the IP of the asterisk - set DEFAULT_AUTHENTIC... to 'asterisk' - removed all STUN entries etc. - provided user name and pwd according to configured SIP friend - SIP communicator registers with asterisk outgoing/incoming call - signalling seems to work, but no audio due to difficulties to find the appropriate codec. I'll give it another try later on... Cheers Jörg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman Sent: Sunday, October 23, 2005 10:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Anyone using Java SIP communicator with Asterisk ? Hi, this java video softphone claims it can operate with Windows messenger. It's also mentioned on this web page http://www.voip-info.org/wiki/view/SIP+COMMUNICATOR But I couldn't find any more info on how to set it up with Asterisk and how compatible is with other video softphones... Anyone with such experience or working installation ? Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Isdntrace utility
On Thu, Oct 20, 2005 at 10:45:38AM +0200, Giordano Grandis wrote: Hi all, i'm looking for an utility that let me trace an ISDN trunk (or all ISDN traffic) on HFC PCI card. ZapHFC is zaptel. You can basically use all the tools availble for Zaptel. What exactly do you want to trace? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7905G Power over Ethernet - does it work?
Perhaps this question should be directed to Cisco support, but since these guys made me nuts (please check that your cable is plugged in correctly etc.), I thought I'd ask here. We bought a Cisco 7905G phone, which boasts to have PoE (Power over Ethernet) support. We have a Netgear FS108P PoE switch, which works with other PoE devices, but not with this Cisco phone. I searched the voip wiki - http://www.voip-info.org/wiki-Cisco+POE - and found a suggestions to reverse some cables in the ethernet wire. So I did, but Cisco 7905G phone still doesn't power up. Does anyone have any suggestions on how to make this phone work with a PoE switch? -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk user meeting in Oslo, Norway
The Norwegian Asterisk user's group is meeting on Tuesday next week. A full one-day seminar in several tracks covering Asterisk is arranged in Oslo. See http://www.asterisk.no for the agenda. I will attend the meeting and enjoy listening to people's experience of Asterisk and various case-studies. If you are in the neighbourhood, make sure that you attend this meeting. /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk user meeting in Sacramento, California
The asterisk users group in Sacramento, California is going to have its first meeting a week from friday, and I would be interested in talking to anyone that is on this list that would think about going. If you are in Sacramento please email me off list. I have a place holder site for now for this meeting at http://www.0xdecafbad.com/Sacramento-Asterisk-Users-Group.html detailing location, etc. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7905G Power over Ethernet - does it work?
Tomasz Chmielewski ha scritto: We bought a Cisco 7905G phone, which boasts to have PoE (Power over Ethernet) support. the 7905 can be powered using pre-standard inline power. So it /doesn't do 802.3af/ I searched the voip wiki - http://www.voip-info.org/wiki-Cisco+POE - and found a suggestions to reverse some cables in the ethernet wire. So I did, but Cisco 7905G phone still doesn't power up. No way to power up the phone is the the switch can be forced to send power in any case. Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Zealand Asterisk Users Group
Hi, Since we're doing this... There is now a New Zealand Asterisk Users Group set up. There is a wiki and mailing list at http://astug.org.nz both are sparse at the moment and could do with some input. If you're in New Zealand (or not) and interested in Asterisk then sign up and get contributing! Thanks, and please excuse the spam. hads -- I can't decide whether to commit suicide or go bowling. -- Florence Henderson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7905G Power over Ethernet - does it work?
Sergio Chersovani ha scritto: No way to power up the phone is the the switch can be forced to send power in any case. I meant that the phone can power up with a custom poe injector that does not care about 802.3af Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7905G Power over Ethernet - does it work?
Sergio Chersovani schrieb: Sergio Chersovani ha scritto: No way to power up the phone is the the switch can be forced to send power in any case. I meant that the phone can power up with a custom poe injector that does not care about 802.3af does poe injector = poe switch (is poe switch and poe injector the same thing but a different name)? if so, it means my switch is not dumb enough or what? anyone knows if it can be dumbified (some special cable, adapter etc.)? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7905G Power over Ethernet - does it work?
Tomasz Chmielewski ha scritto: if so, it means my switch is not dumb enough or what? yes. And the cisco pre-standard poe has reverse pinouts. I guess your switch does not send power because it doesn't see that the cisco phone wants power. I dunno if netgear can force the power injection. Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agent logout
Hi, is there an Agentlogout procedure opposite of the one we get with Agentlogin ? I tried simply having another agent log from the same extension, but when I try Show agents 10 (Alessio) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 51 (Giuliano) available at '[EMAIL PROTECTED]' (musiconhold is 'default') So another question could be: to who calls are counted if answered ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7905G Power over Ethernet - does it work?
Hello Tomasz, I got the 7905 working with an Dell POE switch without any modifications of cables, the 7960 also works on the Dell switch but you have to modify the cable. I also tried the Netgear FS108p and it does not work with the 7905, 7912 and 7960 as I have tested. Even with modified cables no go on the Netgear. I believe the Cisco uses the CDP protocol to get juice from the switch, and the Netgear doesn't understand that. Regards, Mus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] connect 2 phones like in FOP
Hello, Is it possible to connect 2 (SIP) phones via the dialplan. Sort of like dragging 2 phones to each other in Flash operator panel. The thing is I need an action in the dialplan that will connect 2 phones to each other as a reaction to an event without any intervention from one of the 2 phones. Regards, Mus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7905G Power over Ethernet - does it work?
rulle mus schrieb: Hello Tomasz, I got the 7905 working with an Dell POE switch without any modifications of cables, the 7960 also works on the Dell switch but you have to modify the cable. I also tried the Netgear FS108p and it does not work with the 7905, 7912 and 7960 as I have tested. Even with modified cables no go on the Netgear. I believe the Cisco uses the CDP protocol to get juice from the switch, and the Netgear doesn't understand that. thanks. could you tell me the model of the Dell POE switch you use? -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinguishing National from International Calls on Zap Channel
Hi there, can anybody tell me how can i distinguish an national from an international call. The CallerID on the channel doesn't have any leading '0' or '00' so that it is possible that i cannot be sure what type of call i have. i have tried to include 'nationalprefix' and 'internationalprefix' to zapata.conf, as proposed in http://www.asteriskguru.com/tutorials/pri_zaptel.html but while restarting * tells me that these options are unknown. Is there any way to access NPI or TON information of an incoming call on Zap Channels? Tobias Wolf ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iaxmodem
Hi all, I am trying to use iaxmodem I defined iax extension (591) and started iaxmodem. iaxmodem registers with asterisk (is on the same box) when a fax calls this extension, i get Registration completed successfully. Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00011ms SCall: 4 DCall: 0 [192.168.1.10:4569] VERSION : 2 CALLED NUMBER : s CALLING NUMBER : 010xxx CALLING NAME: 010xxx LANGUAGE: it USERNAME: 591 FORMAT : 4 CAPABILITY : 63492 ADSICPE : 0 DATE TIME : 190407242 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00011ms SCall: 30757 DCall: 4 [192.168.1.10:4569] Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 1ms SCall: 30757 DCall: 4 [192.168.1.10:4569] CAUSE : No matching codec support In iax.conf I put: [general] language=it bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all ;allow=gsm allow=ulaw allow=alaw ;allow=h261 ; H.261 ci fa buono (pare...) ;allow=h263 ; H.263 is our video codec mailboxdetail=yes #include iax_additional.conf #include iax_custom.conf In iax_additional.conf I put:[591] username=591 type=friend secret=password record_out=On-Demand record_in=On-Demand qualify=no notransfer=yes host=dynamic disallow=all context=from-internal callerid=Fax Frame 591 allow=ulaw Actually I don't undertand which is the missing codec ( No matching codec support) Next step will be to try to setup the Hylafax server (I already installed it on the same pc and confgured to use the iaxmodem, copying and adapting the template provided with iaxmodem into /var/spool/hylafax/etc/config thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Agent logout
Hi Alessio, The opposite of logging in with Agentlogin is simply hanging up the phone! :-) If you use AgentCallBack, you can instead logoff explicitly. You vcan also log off users manually from the console. Hope this helps l. On Tue, 25 Oct 2005 12:00:07 +0200, Alessio Focardi [EMAIL PROTECTED] wrote: Hi, is there an Agentlogout procedure opposite of the one we get with Agentlogin ? I tried simply having another agent log from the same extension, but when I try Show agents 10 (Alessio) available at '[EMAIL PROTECTED]' (musiconhold is 'default') 51 (Giuliano) available at '[EMAIL PROTECTED]' (musiconhold is 'default') So another question could be: to who calls are counted if answered ? Tnx for any help! -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
On Tue, 2005-10-25 at 12:29 +0200, [EMAIL PROTECTED] wrote: Hi all, I am trying to use iaxmodem Great. It appears to solve a real problem in a very cost effective mannor and needs people playing with it to find any bugs that havent been found yet. Timestamp: 1ms SCall: 30757 DCall: 4 [192.168.1.10:4569] CAUSE : No matching codec support It only supports slinear afaik unless you play with the defines during the build process. Does the account on your asterisk box have slinear enabled (looking at the conf no) or did you follow the directions to enable ulaw/alaw (think its only ulaw but its been over a week since I looked at that part). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7905G Power over Ethernet - does it work?
Dell 3424P, has poe, Qos,and Vlan Mus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Format of extensions.conf
On Tue, 2005-10-25 at 00:52 -0400, Leif Madsen wrote: Now, as someone has also pointed out, using quotes around the string is probably better form as it should handle spaces and such. In expressions only. Set() command is broken in this area (1.2beta and CVS HEAD). To clear, for example, calleridname one must write Set(CALLERID(name)=) The command Set(CALLERID(name)=) will set the name part of callerid to guess what?-) Yes, to a string containing 2 double quote characters! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
Ok, I am sorry, I didn't understand the slinear codec option now I added the slinear codec into the 591 definition [591] username=591 type=friend secret=password record_out=On-Demand record_in=On-Demand qualify=no notransfer=yes host=dynamic disallow=all context=from-internal callerid=Fax Frame 591 allow=slinear something changed, but the problem moved on. Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 6ms SCall: 21132 DCall: 2 [192.168.1.10:4569] Incoming call connected s, 010yyy, 010yyy. Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT Timestamp: 00025ms SCall: 21132 DCall: 2 [192.168.1.10:4569] FORMAT : 64 Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass: RINGING Timestamp: 3ms SCall: 21132 DCall: 2 [192.168.1.10:4569] Unable to pass the full buffer onto the device file. 2015 bytes of 2048 written.Unable to pass the full buffer onto the device file. -1 bytes of 1 written.Oct 25 12:48:39.61: XOFF Unable to pass the full buffer onto the device file. -1 bytes of 1 written.Oct 25 12:48:39.61: XON, 2048 bytes available Unable to pass the full buffer onto the device file. -1 bytes of 2048 written.Unable to pass the full buffer onto the device file. -1 bytes of 1 written.Oct 25 12:48:39.61: XOFF Unable to pass the full buffer onto the device file. -1 bytes of 1 written.Oct 25 12:48:39.61: XON, 2048 bytes available Unable to pass the full buffer onto the device file. -1 bytes of 1982 written.Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00025ms SCall: 2 DCall: 21132 [192.168.1.10:4569] and so on so it seems I am having trouble with device file, I think: Setting device = '/dev/ttyIAX' actually I don't have any '/dev/ttyIAX' Who should have created it ? How can I create it ? thanks in advance,Andrea trixter aka Bret McDanel [EMAIL PROTECTED] To ad.com Asterisk Users Mailing List - Sent by: Non-Commercial Discussion asterisk-users-bo asterisk-users@lists.digium.com [EMAIL PROTECTED] cc m.com Subject Re: [Asterisk-Users] iaxmodem 25/10/2005 12.34 Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com On Tue, 2005-10-25 at 12:29 +0200, [EMAIL PROTECTED] wrote: Hi all, I am trying to use iaxmodem Great. It appears to solve a real problem in a very cost effective mannor and needs people playing with it to find any bugs that havent been found yet. Timestamp: 1ms SCall: 30757 DCall: 4 [192.168.1.10:4569] CAUSE : No matching codec support It only supports slinear afaik unless you play with the defines during the build process. Does the account on your asterisk box have slinear enabled (looking at the conf no) or did you follow the directions to enable ulaw/alaw (think its only ulaw but its been over a week since I looked at that part). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 (See attached file: signature.asc) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: Binary data ___ --Bandwidth and Colocation
[Asterisk-Users] Realtime sip register=
i want to put sip peer registration command register = in my database . anybody have any idea about it how to do this fahd ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
On Tue, 2005-10-25 at 12:57 +0200, [EMAIL PROTECTED] wrote: so it seems I am having trouble with device file, I think: Setting device = '/dev/ttyIAX' actually I don't have any '/dev/ttyIAX' Who should have created it ? How can I create it ? thanks in advance,Andrea At least it appears to be progress :) iaxmodem.c will create a device and symlink /dev/ttyIAX to it, or it should anyway. It does an openpty() to create the device entry. This should just work. Does iaxmodem run with enough privs to write to /dev (most likely it needs to be root) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
On Tue, 2005-10-25 at 04:11 -0700, trixter aka Bret McDanel wrote: On Tue, 2005-10-25 at 12:57 +0200, [EMAIL PROTECTED] wrote: so it seems I am having trouble with device file, I think: Setting device = '/dev/ttyIAX' actually I don't have any '/dev/ttyIAX' Who should have created it ? How can I create it ? thanks in advance,Andrea At least it appears to be progress :) iaxmodem.c will create a device and symlink /dev/ttyIAX to it, or it should anyway. It does an openpty() to create the device entry. This should just work. Does iaxmodem run with enough privs to write to /dev (most likely it needs to be root) I wanted to correct myself, I said 'create' and it doesnt actualy create it, it just attaches to an existing unused one. Figured I should catch it before someone else does :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Swissvoice Vizufon firmware
Hello, Does anybody know when I can find firmware for Swissvoice Vizufon (CIP-5500)? Google isn't sayng anything... I want to update firmware because when I call somebody, Asterisk says: WARNING[22728]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED]' sip.conf is: [xx] type=friend host=dynamic qualify=yes callerid=asd fromuser=xx username=xx secret=pass context=somecontext In phone I have user name and auth. id set to xx and password to pass. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
Now I have the file /dev # dir ttyIAX lrwxrwxrwx1 root root 10 Oct 25 13:52 ttyIAX - /dev/pts/1 The file is now created when I start iasxmodem, and deleted when I quit the app. the result is the same, it is not able to write to the device I am root, iaxmodem is running as root, asterisk is running as asterisk Can I try to run iaxmodem as asterisk too, and change the owner to some libraries from root to asterisk ? Andrea trixter aka Bret McDanel [EMAIL PROTECTED] To ad.com Asterisk Users Mailing List - Sent by: Non-Commercial Discussion asterisk-users-bo asterisk-users@lists.digium.com [EMAIL PROTECTED] cc m.com Subject Re: [Asterisk-Users] iaxmodem 25/10/2005 13.15 Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com On Tue, 2005-10-25 at 04:11 -0700, trixter aka Bret McDanel wrote: On Tue, 2005-10-25 at 12:57 +0200, [EMAIL PROTECTED] wrote: so it seems I am having trouble with device file, I think: Setting device = '/dev/ttyIAX' actually I don't have any '/dev/ttyIAX' Who should have created it ? How can I create it ? thanks in advance,Andrea At least it appears to be progress :) iaxmodem.c will create a device and symlink /dev/ttyIAX to it, or it should anyway. It does an openpty() to create the device entry. This should just work. Does iaxmodem run with enough privs to write to /dev (most likely it needs to be root) I wanted to correct myself, I said 'create' and it doesnt actualy create it, it just attaches to an existing unused one. Figured I should catch it before someone else does :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 (See attached file: signature.asc) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: Binary data ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
On Tue, 2005-10-25 at 14:00 +0200, [EMAIL PROTECTED] wrote: Now I have the file /dev # dir ttyIAX lrwxrwxrwx1 root root 10 Oct 25 13:52 ttyIAX - /dev/pts/1 That should be about right, there are various differences in systems but generally that should work for linux. The file is now created when I start iasxmodem, and deleted when I quit the app. the result is the same, it is not able to write to the device I am root, iaxmodem is running as root, asterisk is running as asterisk Ok, iaxmodem probably needs root to unlink and symlink the /dev/ttyIAX file, given /dev settings. Or at least be suid until after it has done this. Can I try to run iaxmodem as asterisk too, and change the owner to some libraries from root to asterisk ? asterisk doesnt access the device itself. Does /dev/pts/1 actually exist? It should, but lets just cover the bases :) Note that /dev/pts/1 is dynamically assigned, it may change, especially if you have others log into that system on a pseudo tty (ie not console, xterms take a pseudo terminal as do ssh logins). It should however, on your system anyway, be /dev/pts/XX where XX is some integer. The fact that its /dev/pts/1 makes me think that something is on /dev/pts/0 already, which means that kernel support for it should be there, unless by some amazing coincidence it was limited to 1 (default is 256 iirc). So I really dont think that is the problem. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pppoe-server Asterisk
Has anyone had any experience of setting up their * server as a pppoe server such that devices would link to the server running * using pppoe and then do SIP over the PPP interface. I sounds simple and workable for specific handsets / IAD's that support pppoe. Stuart ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime sip register=
You could check these links. I'm trying to do the sip peer registration like this but I get some error about username / auth name mismatch. I think I do something wrong in the MySQL table. I hope it works for you and if it works I would like to hear it from you. Good luck http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime http://www.voip-info.org/wiki/view/Asterisk+RealTime+SipOn 10/25/05, Fahd [EMAIL PROTECTED] wrote:i want to put sip peer registrationcommand register = in my database . anybody have any idea about it how to do thisfahd___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tijmen van den BrinkWilhelminaweg 463441 XC WoerdenTel: 0642233831MSN: [EMAIL PROTECTED]Skype: [EMAIL PROTECTED]SIP:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
asterisk01:/dev # dir /dev/pts/ total 93 drwxr-xr-x2 root root0 Oct 13 20:51 . drwxr-xr-x 30 root root94840 Oct 25 13:52 .. crw--w1 root tty 136, 0 Oct 17 13:17 0 crw--w1 root tty 136, 1 Oct 25 13:52 1 crw--w1 root tty 136, 2 Oct 25 14:23 2 crw--w1 root tty 136, 3 Oct 25 14:22 3 crw--w1 root tty 136, 4 Oct 25 14:21 4 so I have /dev/pts/1 If I quit iaxmodem (CTRL+C) asterisk01:/dev # dir /dev/pts/ total 93 drwxr-xr-x2 root root0 Oct 13 20:51 . drwxr-xr-x 30 root root94816 Oct 25 14:23 .. crw--w1 root tty 136, 0 Oct 17 13:17 0 crw--w1 root tty 136, 2 Oct 25 14:23 2 crw--w1 root tty 136, 3 Oct 25 14:23 3 crw--w1 root tty 136, 4 Oct 25 14:21 4 so /dev/pts/1 has gone away. It seems to be OK, at least to me. Here is a new trial. The first time it seems to write something (2015 of 2048 bytes) Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 7ms SCall: 1 DCall: 0 [192.168.1.10:4569] VERSION : 2 CALLED NUMBER : s CALLING NUMBER : 010xxx CALLING NAME: 010xxx LANGUAGE: it USERNAME: 591 FORMAT : 64 CAPABILITY : 63552 ADSICPE : 0 DATE TIME : 190411589 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 7ms SCall: 04700 DCall: 1 [192.168.1.10:4569] Incoming call connected s, 010xxx, 010xxx. Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT Timestamp: 6ms SCall: 04700 DCall: 1 [192.168.1.10:4569] FORMAT : 64 Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass: RINGING Timestamp: 3ms SCall: 04700 DCall: 1 [192.168.1.10:4569] Unable to pass the full buffer onto the device file. 2015 bytes of 2048 written.Unable to pass the full buffer onto the device file. -1 bytes of 1 written.Oct 25 14:26:11.49: XOFF Unable to pass the full buffer onto the device file. -1 bytes of 1 written.Oct 25 14:26:11.49: XON, 2048 bytes available Unable to pass the full buffer onto the device file. -1 bytes of 2048 written.Unable to pass the full buffer onto the device file. -1 bytes of 1 written.Oct 25 14:26:11.49: XOFF Unable to pass the full buffer onto the device file. -1 bytes of 1 written.Oct 25 14:26:11.49: XON, 2048 bytes available Unable to pass the full buffer onto the device file. -1 bytes of 1982 written.Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 6ms SCall: 1 DCall: 04700 [192.168.1.10:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 1 DCall: 04700 [192.168.1.10:4569] Tx-Frame Retry[010] -- OSeqno: 002 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 02005ms SCall: 04700 DCall: 1 [192.168.1.10:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 003 Type: IAX Subclass: PONG Timestamp: 02005ms SCall: 1 DCall: 04700 [192.168.1.10:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 02005ms SCall: 04700 DCall: 1 [192.168.1.10:4569] Unable to pass the full buffer onto the device file. 2015 bytes of 2048 written.Unable to pass the full buffer onto the device file. -1 bytes of 1 written.Oct 25 14:26:16.50: XOFF Unable to pass the full buffer onto the device file. -1 bytes of 1 written.Oct 25 14:26:16.50: XON, 2048 bytes available Unable to pass the full buffer onto the device file. -1 bytes of 2048 written.Unable to pass the full buffer onto the device file. -1 bytes of 1 written.Oct 25 14:26:16.50: XOFF Unable to pass the full buffer onto the device file. -1 bytes of 1 written.Oct 25 14:26:16.50: XON, 2048 bytes available Unable to pass the full buffer onto the device file. -1 bytes of 1982 written.Unable to pass the full buffer onto the device file. -1 bytes of 2 written.Unable to pass the full buffer onto the device file. -1 bytes of 4 written.Unable to pass the full buffer onto the device file. -1 bytes of 2 written.Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10023ms SCall: 1 DCall: 04700 [192.168.1.10:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 10023ms SCall: 04700 DCall: 1 [192.168.1.10:4569] Tx-Frame Retry[010] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: LAGRP Timestamp: 10023ms SCall: 04700 DCall: 1 [192.168.1.10:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 10023ms SCall: 1 DCall: 04700 [192.168.1.10:4569] Tx-Frame Retry[010] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass: PING Timestamp: 12015ms SCall: 04700 DCall: 1
Re: [Asterisk-Users] iaxmodem
I am sorry, I didn't mention another problem. I WAS NOT able to compile the spandsp lib shipped with iaxmodem. ./configure is OK makeinstead returns asterisk01:/usr/src/iaxmodem-0.0.5/lib/spandsp # make Making all in src make[1]: Entering directory `/usr/src/iaxmodem-0.0.5/lib/spandsp/src' cd .. /bin/sh /usr/src/iaxmodem-0.0.5/lib/spandsp/config/missing --run autoheader Can't locate object method path via package Autom4te::Request at /usr/bin/autom4te line 81. autoheader: autom4te failed with exit status: 1 at /usr/bin/autoheader line 163 make[1]: *** [../config-h.in] Error 1 make[1]: Leaving directory `/usr/src/iaxmodem-0.0.5/lib/spandsp/src' make: *** [all-recursive] Error 1 I have already installed spandsp, I already use spandsp for app_rxfax and app_txfax on that box The spandsp package I am using are the last ones, spandsp-0.0.2pre21.tar.gz Andrea ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
On Tue, 2005-10-25 at 14:29 +0200, [EMAIL PROTECTED] wrote: Timestamp: 3ms SCall: 04700 DCall: 1 [192.168.1.10:4569] Unable to pass the full buffer onto the device file. 2015 bytes of 2048 written.Unable to pass the full buffer onto the device file. -1 bytes of 1 written.Oct 25 14:26:11.49: XOFF -1 indicates error and perror() should be called (or something that will be more meaningful). That is something that potentially should be added to iaxmodem in the near future. Are you running anything that will read /dev/ttyIAX like hylafax? If there is nothing that can read the device the buffer for that device will become full and you will see these errors. You may need to install hylafax now and get that running. The README that comes with iaxmodem states there is a modem entry for hylafax and how to integrate that. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Park question
Hello: We are using Asterisk as a voicemail and media server. Call processing is done by a different box running SER. I am experiencing a problem when trying to implement call park on Asterisk. The call is transferred to the parking lot OK but parkandannounce wants to dial the calling party to announce the lot number and this fails because the calling party is busy. I'm wondering if other people have experienced this problem and if so how did you address it? Thanks -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
Probably you are right I installed hylafax and configured it to use iaxmodem, but I didn't start it Now I will research how to start hylafax, and I will try again Andrea trixter aka Bret McDanel [EMAIL PROTECTED] To ad.com Asterisk Users Mailing List - Sent by: Non-Commercial Discussion asterisk-users-bo asterisk-users@lists.digium.com [EMAIL PROTECTED] cc m.com Subject Re: [Asterisk-Users] iaxmodem 25/10/2005 14.36 Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com On Tue, 2005-10-25 at 14:29 +0200, [EMAIL PROTECTED] wrote: Timestamp: 3ms SCall: 04700 DCall: 1 [192.168.1.10:4569] Unable to pass the full buffer onto the device file. 2015 bytes of 2048 written.Unable to pass the full buffer onto the device file. -1 bytes of 1 written.Oct 25 14:26:11.49: XOFF -1 indicates error and perror() should be called (or something that will be more meaningful). That is something that potentially should be added to iaxmodem in the near future. Are you running anything that will read /dev/ttyIAX like hylafax? If there is nothing that can read the device the buffer for that device will become full and you will see these errors. You may need to install hylafax now and get that running. The README that comes with iaxmodem states there is a modem entry for hylafax and how to integrate that. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 (See attached file: signature.asc) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: Binary data ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question on callingpres and blocked numbers
Hi, Does anyone know what the legalness is of unblocking a blocked call? For instance, when someone blocks their number it comes into our system with the block flag (across PRI). It is then passed on to the ATA as blocked. Is it legal for me to set the flag back to unblock the call? (I realize no one here is probably a lawer but was just curious to see what others thought). I can't think of anything that would be illegal with it, perhaps unethical. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
On Tue, 2005-10-25 at 14:36 +0200, [EMAIL PROTECTED] wrote: I am sorry, I didn't mention another problem. I WAS NOT able to compile the spandsp lib shipped with iaxmodem. I am unsure what comes with iaxmodem, I do know that the iaxmodem project has fixed some spandsp problems and those patches have been given to the spandsp team (which I think is just one guy) for integration. I dont know if all of them have been integrated, so you may see some performance issues ... For those problems Lee would be a better person to talk to, I am not familiar enough with what it is trying to do to answer and dont have the time to dig right now. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MWI for other purpose than voicemail?
Hi, I.e. I want to be able to turn on and off the MWI light independant of any voicemail function in asterisk. Is this at all possible? (We have Polycom IP300 phones, and the MWI light works fine with voicemail and SendText()). /Ola ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FCT-11M
Thank you. After some reboots and repeated testing, I've refined my observations. The no-audio problem is gone (no explanation). Through further experimentation I've been able to observe a few consistent things about behavior in its current condition... The main problem seems to be related to disconnect signalling. Simply put, the channel didn't hang up after the GSM connection ends. Instead, I would hear a dial tone thru the bridged side of the call (it's not a North American dialtone). I considered changing the zone in indications.conf, but that is system-wide, and probably inappropriate because I have a Verizon POTS line too. Then I tried putting hanguponpolarityswitch=yes in zapata.conf and it worked - the call would be torn down. But... ...that introduced a new problem on outbound calling, because the channel would hangup immediately upon remote answer. So I added answeronpolarityswitch=yes too, but it had no effect. I also messed around with polarityonanswerdelay= but I was operating in the dark and it didn't help. So I tried to load modules with debug options (zaptel, wctdm, wcfxo), hoping to see more info about device behavior in realtime, but I see nothing new in the message log. But I'm kind of bumbling and stumbling on that. If anyone can offer more precise guidance, I'd be grateful. - Date: Mon, 24 Oct 2005 22:24:21 -0700 From: OTR Comm [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] GSM gateway for Asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I forwarded your note below to [EMAIL PROTECTED] I found some docas on the FCT-11M at their site, but it was in Chinese, so I sent them your problem. Hope they will respond to this list and maybe to you directly. Murrah Boswel - Original Message - From: Bill Michaelson [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, October 24, 2005 9:42 PM Subject: [Asterisk-Users] GSM gateway for Asterisk I recently obtained a FCT-11M GSM-analog converter box. It arrived with no documentation. So I popped in a SIM chip, and connected the the RJ11 port to an FXO port on my Asterisk box. It worked smoothly right away for inbound and outbound calls in all respects. For about an hour. Then either spontaneously or due to some action I've been unable to identify, call supervision and other functions became flaky. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pppoe-server Asterisk
Stuart Hirst wrote: Has anyone had any experience of setting up their * server as a pppoe server such that devices would link to the server running * using pppoe and then do SIP over the PPP interface. I sounds simple and workable for specific handsets / IAD's that support pppoe. Stuart ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, I have three clients running asterisk pbx servers with pppoe client (Verizon DSL) with no problems (small companys with 5-10 phones). Best regards, Chris HARIGA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
I realize that on my box was not installed vgetty and agetty ( i think that they are demanded from hylafax to get data from the ttyIAX device) I have added them, reconfigured re-make and re-installed hylafax and restarted it. The problem, now, is about egetty, which actually dos not exists. Moreover, running ./configure no ask about egetty, but only about agetty and vgetty I see on the net that is an old problem, now I will reserch how to fix it Andrea trixter aka Bret McDanel [EMAIL PROTECTED] To ad.com Asterisk Users Mailing List - Sent by: Non-Commercial Discussion asterisk-users-bo asterisk-users@lists.digium.com [EMAIL PROTECTED] cc m.com Subject Re: [Asterisk-Users] iaxmodem 25/10/2005 14.51 Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com On Tue, 2005-10-25 at 14:36 +0200, [EMAIL PROTECTED] wrote: I am sorry, I didn't mention another problem. I WAS NOT able to compile the spandsp lib shipped with iaxmodem. I am unsure what comes with iaxmodem, I do know that the iaxmodem project has fixed some spandsp problems and those patches have been given to the spandsp team (which I think is just one guy) for integration. I dont know if all of them have been integrated, so you may see some performance issues ... For those problems Lee would be a better person to talk to, I am not familiar enough with what it is trying to do to answer and dont have the time to dig right now. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 (See attached file: signature.asc) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: Binary data ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Siemens HI-path to ASTERISK
On Mon, Oct 24, 2005 at 06:42:02PM -0200, [EMAIL PROTECTED] wrote: Yes, for sure, it works. With TE110P from Digium using E1/ISDN/Pri signalling. By heart, I remember the following: 1. Configure Siemens E1 port as station and Asterisk as Pri_Net (or Central Office). 2. At Siemens, set the E1 port as S2 Point-to-Point net line without CRC4 or something like this. yep done. i only have a problem i can call any extension in the pbx but i can't take outside line with the 9 you can send to me the extensions.conf please please/ 3. At Asterisk, put these lines (/etc/zaptel.conf): span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 You have to study the rest of * conf file, but these ones are the important ones. Regards, --hg - Original Message - From: Pablo Allietti [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, October 24, 2005 6:55 PM Subject: [Asterisk-Users] Siemens HI-path to ASTERISK anybody can connect a Siemens HI-PATH to ASterisk via e1 ? i need your help please. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Realtime sip register=
Hello! As I know, the "register" is a variable of [general] section in sip.conf. You can't use it in database, ie you can't add new registers without reload the asterisk.. I am right? Regards. Jsalas. -Mensaje original-De: tijmen van den brink [mailto:[EMAIL PROTECTED]Enviado el: Tuesday, October 25, 2005 9:26 AMPara: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: Re: [Asterisk-Users] Realtime sip register=You could check these links. I'm trying to do the sip peer registration like this but I get some error about username / auth name mismatch.I think I do something wrong in the MySQL table.I hope it works for you and if it works I would like to hear it from you.Good luckhttp://www.voip-info.org/tiki-index.php?page=Asterisk+RealTimehttp://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip On 10/25/05, Fahd [EMAIL PROTECTED] wrote: i want to put sip peer registrationcommand register = in my database . anybody have any idea about it how to do thisfahd___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tijmen van den BrinkWilhelminaweg 463441 XC WoerdenTel: 0642233831MSN: [EMAIL PROTECTED]Skype: [EMAIL PROTECTED]SIP:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Redundency
Benjamin Lawetz wrote:
Since I can't do that, what I've settled on is heartbeat + mon.
Heartbeat will monitor for a system level failure and switch to the backup
machine if neccesary; and mon will watch the asterisk (or any
other) service and restart it and/or alert me if it fails.
What kind of monitor are you using to monitor asterisk?
Sorry for my slow response. My asterisk monitor right now is
embarrassingly simple. All it does is execute show uptime and look for
output starting with System, see below. Obviously the method has
limitations. 1) It will only really only tell me that the daemon is
running, not that it's able to carry any calls. 2) It only works on
localhost.
Input on how to test a remote instance of asterisk would be welcome, as
well as a method of making a test call or reliably testing for the
ability to make calls. My impression is that this would require
asterisk to have a Dial command in the CLI, or a linux SIP client that
I could execute from the shell. I'm not aware of the existence of either.
Any other simple and reliable methods of testing asterisk's condition
would be welcome.
The alerts, by the way are pretty simple as well. See the excerpt from
mon.cf below. restartasterisk.alert does exactly what it says.
stopeverything.alert shuts down heartbeat, which will cause another node
in the cluster to take over...in fact that node will start mon, which
will then use the restartasterisk.alert to start up asterisk. Asterisk
only starts on the backup machine when the primary fails so that config
changes replicated from the primary will take effect. Total downtime
should be 3min. Which will let me hit 5-nine if it only happens once
a year ;)
Config changes are replicated via rsync and ssh every few minutes.
Voicemails are also copied from primary to backup by rsync. One thing I
still need to do is make rsync stop attempting to replicate files when
the failover occurrs. That will probably just require another alert
below the stopeverything.alert.
The replication of couse means that this setup will not protect me from
a bad config change that breaks asterisk, as that change will be
replicated throughout the cluster. So all significant config changes
should be tested on a standalone box.
[EMAIL PROTECTED] mon]# cat /usr/lib/mon/mon.d/asterisk.monitor
#!/bin/sh
##can only check localhost. Always checks localhost regardless of input
SHOW_UPTIME=`/usr/sbin/asterisk -rx show uptime | /bin/cut -b 1-6`
if [ $SHOW_UPTIME == System ]; then
exit 0
else
echo localhost
exit 1
fi
From mon.cf:
watch asterisk
service asterisk
description asterisk pbx on localhost
interval 10s
monitor asterisk.monitor
period wd {Sun-Sat}
alert mail.alert [EMAIL PROTECTED]
alert restartasterisk.alert [EMAIL PROTECTED]
alertevery 30s
service asterisk-failover
description checking if we need to stop heartbeat
interval 10s
monitor asterisk.monitor
period wd {Sun-Sat}
alert stopeverything.alert [EMAIL PROTECTED]
alertafter 5 3m
___
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Re: [Asterisk-Users] Realtime sip register=
Dear Juan I think you are right. you can't add new registers without reload asterisk. and the register can't be put in the REALTIME database. But there is an alternative to put the sip.conf file in in the database. This is a bit different from the REALTIME engine. This is just a database table with the sip.conf entries. I think you still need to reload if you change the table entries. Extract from /etc/asterisk/extconfig.conf ; Static configuration files: ; ; file.conf = driver,database[,table] ; ; maps a particular configuration file to the given ; database driver, database and table (or uses the ; name of the file as the table if not specified) ; The following files CANNOT be loaded from Realtime storage: ; asterisk.conf ; extconfig.conf (this file) ; logger.conf Are Casilla -- http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk and Drupal Consultants http://astbill.com - Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com On 10/25/05, Juan Salas [EMAIL PROTECTED] wrote: Hello! As I know, the register is a variable of [general] section in sip.conf. You can't use it in database, ie you can't add new registers without reload the asterisk.. I am right? Regards. Jsalas. -Mensaje original-De: tijmen van den brink [mailto:[EMAIL PROTECTED]]Enviado el: Tuesday, October 25, 2005 9:26 AMPara: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: Re: [Asterisk-Users] Realtime sip register=You could check these links. I'm trying to do the sip peer registration like this but I get some error about username / auth name mismatch.I think I do something wrong in the MySQL table.I hope it works for you and if it works I would like to hear it from you.Good luckhttp://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip On 10/25/05, Fahd [EMAIL PROTECTED] wrote: i want to put sip peer registrationcommand register = in my database . anybody have any idea about it how to do thisfahd___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tijmen van den BrinkWilhelminaweg 463441 XC WoerdenTel: 0642233831MSN: [EMAIL PROTECTED]Skype: [EMAIL PROTECTED] SIP:[EMAIL PROTECTED] ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
I'll see what I can do about improving the error messages involved with the write error, but that's not going to help your problem here. Until the IAXmodem documentation says otherwise, you *must* install and use the spandsp version that ships with IAXmodem. The only exception to this of which I know is that IAXmodem-0.0.5 will work with spandsp-0.0.3pre4 unaltered. The reason for this is because some of the IAXmodem development causes changes (fixes and enhancements) in the spandsp library, specifically the T.31 modem part. Eventually I expect these developments to slow down enough that IAXmodem will work with any version of spandsp after that point, but that point has not yet arrived. My guess is that the errors you're seeing are due to your not using the correct version of spandsp. As for the compilation error you're having with Autom4te (Can't locate object method path via package Autom4te::Request at /usr/bin/autom4te line 81.), I'm not sure what the right answer is. It's probably a dependency problem of some kind. So fix the spandsp build problem and I'll expect that your IAXmodem problem will go away... if not, then let me know. The IAXmodem forums or tracker is probably the better place for that, though. Lee. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival() works fine when I call from PSTN and not when I call from XLite... What's going on?
Hello there, For some reason, Festival() works fine when I call from PSTN (via an IAX connection that I've got from Voice Pulse), but does not produce any sound when I call from my X-Lite SIP phone. However, if I use text2wave instead of Festival(), both my PSTN and my X-Lite connections seem to work fine. Does anyone know what is going on? Note that the Festival server is being called in both cases without any noticeable problems. Thanks in advance for any lead, Leo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where is the text of the voicemail email ??
On Mon, October 24, 2005 9:00 am, Tom Rymes wrote: I would like to be able to edit the pager notification e-mail. ... Looks like this new feature is already checked into CVS. -- Paul Dugas, Computer Engineer Dugas Enterprises, LLC [EMAIL PROTECTED] phone: 404-932-1355 522 Black Canyon Park http://dugas.cc fax: 866-751-6494 Canton, GA 30114 USA -- Onsite at GDOT W.Annex 404-463-2860 x199 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SER for dummies ?
Good Question. We have tested it with any combination we can think about and it is working safely. There is no way (we know about) that you can pass toll free calls. :-) Basically SER is configured to only accept clients that have the same callerid as account numbers so SER refuse to pass the call if you try to be smart. Asterisk only passes the call if you have a valid account and the request is handed over from the SER server. Asterisk determine the max length of the call based on the Users Account balance in AstBill. Are Casilla -- http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk and Drupal Consultants http://astbill.com - Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com On 10/25/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Tue, 2005-10-25 at 08:27 +0100, Are wrote: The authentication in Asterisk is done using ANI/CLI.Same way as broadvoice, wonder if using that setup if I set my caller idto someone else will it cause the INVITE that broadvoice does (broadvoice will invite the person registered as that account if you tryto make a call on their CID, asterisk ignores that invite, I am not sosure if all devices will)--Trixter http://www.0xdecafbad.com Bret McDanelUK +44 870 340 4605 Germany +49 801 777 555 3402US +1 360 207 0479 or +1 516 687 5200FreeWorldDialup: 635378-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.1 (GNU/Linux)iD8DBQBDXeNg+1olxlzQw5cRApWJAJ4sXCutFLLuAk26jzumrS/ioMiZ3ACfa8zZIBWJRwuEQ1RN9EqRvajQG/c==DzJ5-END PGP SIGNATURE-___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AudioCodes - TP260
Title: AudioCodes - TP260 Hi All, Does anyone have any experience with using Asterisk with AudioCodes TP260 SIP board? If yes, please let me know if you have had any problems. Regards, Chard Johnston ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime sip register=
Juan Salas wrote: Hello! As I know, the register is a variable of [general] section in sip.conf. You can't use it in database, ie you can't add new registers without reload the asterisk.. You can have a static config in a database, but you will still have to reload. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
Thanks for the answer. The problem went away starting faxgetty, I am sorry, I didn't carefully read the README Now I have another problem, which probably is exactly what Lee said, a spandsp version error. Now I am trying with the spandsp-0.0.3pre4 version. Andrea Lee Howard [EMAIL PROTECTED] van.com To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 25/10/2005 15.59 Re: [Asterisk-Users] iaxmodem Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com I'll see what I can do about improving the error messages involved with the write error, but that's not going to help your problem here. Until the IAXmodem documentation says otherwise, you *must* install and use the spandsp version that ships with IAXmodem. The only exception to this of which I know is that IAXmodem-0.0.5 will work with spandsp-0.0.3pre4 unaltered. The reason for this is because some of the IAXmodem development causes changes (fixes and enhancements) in the spandsp library, specifically the T.31 modem part. Eventually I expect these developments to slow down enough that IAXmodem will work with any version of spandsp after that point, but that point has not yet arrived. My guess is that the errors you're seeing are due to your not using the correct version of spandsp. As for the compilation error you're having with Autom4te (Can't locate object method path via package Autom4te::Request at /usr/bin/autom4te line 81.), I'm not sure what the right answer is. It's probably a dependency problem of some kind. So fix the spandsp build problem and I'll expect that your IAXmodem problem will go away... if not, then let me know. The IAXmodem forums or tracker is probably the better place for that, though. Lee. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail prompts not heard on Cisco Phone
Got a test setup with CCM 4.1 and Asterisk running kind of successfully. Been trying out using * fro the VM system. I can make calls from the CCM side across to * and answer them using a copy of Xten Lite. If I allow a call to head to voicemail, I can't hear any prompts from the system. If I watch a debug and wait for the recording to start I can leave a message successfully, and if I head to the INBOX directory of the * extension, I can listen to the message wav file. I'm calling across to * from CCM using a 7940 registered with the CCM Server. Anyone got any suggestions on what might be going wrong here. If I listen to the voicemail prompts using the Xten lite softphone, I can hear everything fine. I've thought that I might possibly need to convert the voice prompts from gsm to u-law, but a convert using sox of vm-intro.gsm to a .au file didn't make any difference. I know someone out there must be running things successfully. Did you see any issue like this, or have to do anything particular to get things to be heard on a 7940/ 7960 phone? Thanks for any suggestions. Nathan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Siemens HI-path to ASTERISK
Hi Pablo! I understood your problem. It is related to Siemens PBX. With this topology, Asterisk is acting as a PSTN Central Office (a Public Central). What you asking is something like this: Asterisk acting as Central Office - HiPath - Public Central Office That is: the SIP devices connected to the Asterisk are not HI-Path's extensions! They seem external terminal/lines. So... You will have to enable, at HiPath, something called Transit or External traffic. In other words, it is a feature that you enable on HiPath allowing traffic between two trunks (the trunk connected to Asterisk and the trunk connected to the PSTN Central Office). Here we had to create a trunk access code. So, if a Asterisk user wants to call the outside number -1234, he/she will dial: 9 + -1234 Asterisk with then route this call to HiPath prefixing the trunk access code, for example, 88. So, asterisk will dial: 88 + -1234 Hope this helps, --hg - Original Message - From: [EMAIL PROTECTED] To: Pablo Allietti [EMAIL PROTECTED] Sent: Tuesday, October 25, 2005 11:52 AM Subject: Re: Siemens HI-path to ASTERISK Hi Pablo! I understood your problem. It is related to Siemens PBX. With this topology, Asterisk is acting as a PSTN Central Office (a Public Central). What you asking is something like this: Asterisk acting as Central Office - HiPath - Public Central Office That is: the SIP devices connected to the Asterisk are not HI-Path's extensions! They seem external terminal/lines. So... You will have to enable, at HiPath, something called Transit or External traffic. In other words, it is a feature that you enable on HiPath allowing traffic between two trunks (the trunk connected to Asterisk and the trunk connected to the PSTN Central Office). Here we had to create a trunk access code. So, if a Asterisk user wants to call the outside number -1234, he/she will dial: 9 + -1234 Asterisk with then route this call to HiPath prefixing the trunk access code, for example, 88. So, asterisk will dial: 88 + -1234 Hope this helps, Huelbe. - Original Message - From: Pablo Allietti [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, October 25, 2005 12:41 PM Subject: Re: Siemens HI-path to ASTERISK On Mon, Oct 24, 2005 at 06:42:02PM -0200, [EMAIL PROTECTED] wrote: Yes, for sure, it works. With TE110P from Digium using E1/ISDN/Pri signalling. By heart, I remember the following: 1. Configure Siemens E1 port as station and Asterisk as Pri_Net (or Central Office). 2. At Siemens, set the E1 port as S2 Point-to-Point net line without CRC4 or something like this. yep done. i only have a problem i can call any extension in the pbx but i can't take outside line with the 9 you can send to me the extensions.conf please please/ 3. At Asterisk, put these lines (/etc/zaptel.conf): span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 You have to study the rest of * conf file, but these ones are the important ones. Regards, --hg - Original Message - From: Pablo Allietti [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, October 24, 2005 6:55 PM Subject: [Asterisk-Users] Siemens HI-path to ASTERISK anybody can connect a Siemens HI-PATH to ASterisk via e1 ? i need your help please. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma A104 errors
I am using a A104 Sangoma card. We are runningasterisk cvs head on our production box.After wanpipe configuration I am receiving the below mentioned error. pri show span looks good as below. pri show span 1 Primary D-channel: 24 Status: Provisioned, In Alarm, Down, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 error on the server: Write to 33 failed: Bad address Short write: 0/5 (Bad address) Any help appreciated. Thank you, ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 REGISTRATION PROBLEM: Gatekeeper '[EMAIL PROTECTED] ' found but failed to register
Hi all
First of all excuse me if i make such a big post, hope
also to write in the right place.
I need to connect my linux/asterisk (10.0.0.252) box
to a Nortel PBX (192.168.1.10) with h323
I'd like to allow some phones to register via sip to
asterisk and
with these to the Nortel PBX wich gives me the
connections to the outside world (phone)
after downloading and compiling the latest asterisk
source from cvs
OpenH323 v1.15.6, PWlib v1.8.7 (Mimas version from
Voxgratia)
and oh323-0.7.3 from
http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.7.3.tar.gz
starting asterisk i get
[4]WrapProcess::Main: Starting...
[1]WrapH323EndPoint::WrapH323EndPoint: Compile-time
libraries OpenH323 v1.15.6, PWlib v1.8.7
[2]WrapperAPI::h323_end_point_create: Endpoint
created.
[3]WrapperAPI::h323_set_options: Setting endpoint
options.
[3]WrapperAPI::h323_set_ports: Setting endpoint port
ranges.
[2]WrapperAPI::h323_removeall_capabilities: Removing
all capabilities.
[3]WrapH323EndPoint::RemoveAllCapabilities: Removing
all capabilities of local endpoint.
[5]WrapH323EndPoint::SetFrames: Setting 20
[5]WrapH323EndPoint::GetFrames: Returning 20
[2]WrapperAPI::h323_set_capability: Inserted
capability G.711-ALaw-64k{hw}
[3]WrapperAPI::h323_set_senduimode: User-input mode
set.
[2]WrapperAPI::h323_set_gk: Configuring gatekeeper.
[3]WrapH323EndPoint::SetGatekeeperTimeToLive:
Gatekeeper registration TTL set at 600 sec
[4]GKRegThread::GKRegThread: Object initialized.
[4]GKRegThread::GKRegThread: Unblock pipe - 20, 21
[3]WrapperAPI::h323_callback_register: Callback
functions installed.
[2]GKRegThread::Main: GK: name [192.168.1.10], zone []
[2]GKRegThread::Main: Failed to register with GK name
[192.168.1.10], zone []
[4]WrapperAPI::h323_get_gk: Checking gatekeeper.
-- Gatekeeper '[EMAIL PROTECTED]'
found but failed to register
RAS Failed registration of with
Nortel_H323_Gatekeeper
i'm wondering three things.
FIRST QUESTION
Am'i right in the idea? is asterisk capable the
realize what i need ?
SECOND QUESTION
the guy working in the telco said me that i can see on
the Nortel pbx the connection attempt
but from 127.0.0.1. By reading the oh323.log i can see
that during the RAS phase my asterisk
send the loopback address
in the following log i can see
rasAddress = 1 entries {
[0]=ipAddress {
ip = 4 octets {
7f 00 00 01
}
port = 10002
}
}
0:00.145 GKRegThread:0816ac30 TCP Appending H.225
transport ip$10.0.0.253:1720 using associated
transport Transport[remote=ip$192.168.1.10:1719
if=ip$127.0.0.1:10001]
THIRD QUESTION
why in the string
RAS Failed registration of with
Nortel_H323_Gatekeeper
after the word of there's only a blank space?
thank you very much for your patience and for your
precious help (i hope)
in the oh323.log
0:00.007 asterisk-oh323 H323Created
endpoint.
0:00.029 H323 Cleaner H323Started
cleaner thread
0:00.029 asterisk-oh323 H323Started
listener Listener[ip$10.0.0.253:1720]
0:00.030 asterisk-oh323 H323Added
capability: G.711-ALaw-64k{hw} 1
0:00.030 asterisk-oh323 H323Added
capability: UserInput/hookflash 2
0:00.030 asterisk-oh323 H323Added
capability: UserInput/basicString 3
0:00.030 asterisk-oh323 H323Added
capability: UserInput/dtmf 4
0:00.030 asterisk-oh323 H323Added
capability: UserInput/RFC2833 5
0:00.054H323 Listener:816a698 H323Awaiting TCP
connections on port 1720
0:00.054H323 Listener:816a698 TCP Waiting on
socket accept on ip$10.0.0.253:1720
0:00.054 GKRegThread:0816ac30 H323UDP Binding to
interface: :::10001
0:00.056 GKRegThread:0816ac30 RAS Authenticator
H235AnnexD_Procedure1no-pwd not active during GRQ
SetCapability negotiation
0:00.056 GKRegThread:0816ac30 RAS Authenticator
CATno-pwd not active during GRQ SetCapability
negotiation
0:00.056 GKRegThread:0816ac30 RAS Authenticator
MD5no-pwd not active during GRQ SetCapability
negotiation
0:00.056 GKRegThread:0816ac30 H225Started
gatekeeper discovery of ip$192.168.1.10
0:00.056 GKRegThread:0816ac30 RAS Searching
interfaces:
127.0.0.1
[00:00:00:00:00:00:00:00:00:00:00:00:00:00:00:01]
00-00-00-00-00-00 (lo)
10.0.0.253
[fe:80:00:00:00:00:00:00:02:01:02:ff:fe:12:02:92]
00-01-02-12-02-92 (eth0)
0:00.056 GKRegThread:0816ac30 RAS Gatekeeper
discovery on interface: 10.0.0.253:10002
0:00.057GkMonitor:816cae0 RAS Background
thread started
0:00.086 GKRegThread:0816ac30 Trans Sending PDU:
gatekeeperRequest {
requestSeqNum = 65022
protocolIdentifier = 0.0.8.2250.0.4
rasAddress = ipAddress {
ip = 4 octets {
0a 18 02 fd
}
port = 10002
Re: [Asterisk-Users] Sangoma A104 errors
Have you tried Asterisk 1.2beta1? does it work under that release? We have been using an a104u with PRIs on 1.2b1 for about 6 weeks now with no problems. MATT--- On 10/25/05, Sharon [EMAIL PROTECTED] wrote: I am using a A104 Sangoma card. We are runningasterisk cvs head on our production box.After wanpipe configuration I am receiving the below mentioned error. pri show span looks good as below. pri show span 1 Primary D-channel: 24 Status: Provisioned, In Alarm, Down, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 error on the server: Write to 33 failed: Bad address Short write: 0/5 (Bad address) Any help appreciated. Thank you, ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Format of extensions.conf
On 10/25/05, Sergey Okhapkin [EMAIL PROTECTED] wrote: On Tue, 2005-10-25 at 00:52 -0400, Leif Madsen wrote: Now, as someone has also pointed out, using quotes around the string is probably better form as it should handle spaces and such. In expressions only. Set() command is broken in this area (1.2beta and CVS HEAD). To clear, for example, calleridname one must write Set(CALLERID(name)=) The command Set(CALLERID(name)=) will set the name part of callerid to guess what?-) Yes, to a string containing 2 double quote characters! Yes, I was speaking of expressions specifically, but thanks for clearing this up. Honestly, I don't think I've ever tried to set a NULL string to a variable with a function... that could probably be filed as a bug, but someone might call it a feature :) -- Leif Madsen - http://www.leifmadsen.com http://www.asteriskdocs.org -- Co-Founder http://www.oreilly.com/catalog/asterisk -- Co-Author ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Format of extensions.conf
Taking in the account poor Asterisk documentation, it's a bug. The bug can be called as a feature, only when it is documented:-) On Tue, 2005-10-25 at 10:47 -0400, Leif Madsen wrote: Set(CALLERID(name)=) will set the name part of callerid to guess what?-) Yes, to a string containing 2 double quote characters! Yes, I was speaking of expressions specifically, but thanks for clearing this up. Honestly, I don't think I've ever tried to set a NULL string to a variable with a function... that could probably be filed as a bug, but someone might call it a feature :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A104 errors
Have you tried contacting Sangoma technical support? They are likely the best equipped to support the card and the alarm you're receiving. On 10/25/05, Sharon [EMAIL PROTECTED] wrote: I am using a A104 Sangoma card. We are runningasterisk cvs head on ourproduction box.After wanpipe configuration I am receiving the below mentioned error.pri show span looks good as below.pri show span 1Primary D-channel: 24Status: Provisioned, In Alarm, Down, ActiveSwitchtype: National ISDNType: CPEWindow Length: 0/7 Sentrej: 0SolicitFbit: 0Retrans: 0Busy: 0Overlap Dial: 0T200 Timer: 1000T203 Timer: 1T305 Timer: 3T308 Timer: 4000T313 Timer: 4000N200 Counter: 3error on the server: Write to 33 failed: Bad addressShort write: 0/5 (Bad address)Any help appreciated.Thank you,___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A104 errors
sangoma tech's cldn't help with that error. Also it did work with asterisk 1.0.3 version wanpipe-beta9-2.3.3 now i'm using wanpipe-beta15-2.3.3 with asterisk cvs head probably someone cldanswer this when wanpipex.conf files are created do they depend on the number of spans or number of cards. Thanks, On 10/25/05, Matt Florell [EMAIL PROTECTED] wrote: Have you tried Asterisk 1.2beta1? does it work under that release? We have been using an a104u with PRIs on 1.2b1 for about 6 weeks now with no problems. MATT--- On 10/25/05, Sharon [EMAIL PROTECTED] wrote: I am using a A104 Sangoma card. We are runningasterisk cvs head on our production box.After wanpipe configuration I am receiving the below mentioned error. pri show span looks good as below. pri show span 1 Primary D-channel: 24 Status: Provisioned, In Alarm, Down, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 error on the server: Write to 33 failed: Bad address Short write: 0/5 (Bad address) Any help appreciated. Thank you, ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
For hylafax to answer a call, you need to use faxgetty.. Add this 2 lines to your /etc/inittab and run init q to force a reload: IAX:2345:respawn:/usr/local/bin/iaxmodem ttyIAX modem:2345:respawn:/usr/sbin/faxgetty ttyIAX Change the paths according to your system. Julian J. M. On 10/25/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Probably you are right I installed hylafax and configured it to use iaxmodem, but I didn't start it Now I will research how to start hylafax, and I will try again Andrea ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A104 errors
The number of spans. If you've got a quad card, you can actually configure 4 different wanpipe interfaces on that card. On 10/25/05, Sharon [EMAIL PROTECTED] wrote: sangoma tech's cldn't help with that error.Also it did work with asterisk 1.0.3 version wanpipe-beta9-2.3.3 now i'm using wanpipe-beta15-2.3.3 with asterisk cvs headprobably someone cldanswer thiswhen wanpipex.conf files are createddo they depend on the number ofspans or number of cards.Thanks, On 10/25/05, Matt Florell [EMAIL PROTECTED] wrote: Have you tried Asterisk 1.2beta1? does it work under that release? We have been using an a104u with PRIs on 1.2b1 for about 6 weeks now with no problems. MATT--- On 10/25/05, Sharon [EMAIL PROTECTED] wrote: I am using a A104 Sangoma card. We are runningasterisk cvs head on our production box.After wanpipe configuration I am receiving the below mentioned error. pri show span looks good as below. pri show span 1 Primary D-channel: 24 Status: Provisioned, In Alarm, Down, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 error on the server: Write to 33 failed: Bad address Short write: 0/5 (Bad address)Any help appreciated. Thank you, ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Siemens HI-path to ASTERISK
On Tue, Oct 25, 2005 at 12:31:41PM -0200, [EMAIL PROTECTED] wrote: Hi Pablo! ok. i do all the changes but now i have this error -- Channel 0/1, span 1 got hangup Oct 25 11:46:40 WARNING[3639]: app_dial.c:416 wait_for_answer: Unable to forward voice -- Hungup 'Zap/1-1' == No one is available to answer at this time -- Executing Playback(SIP/205-0014, invalid) in new stack -- Playing 'invalid' (language 'en') == Spawn extension (from-internal, 9122, 2) exited non-zero on 'SIP/205-0014' maybe is a extensions.conf ?? can you paste your extensions.conf here please? I understood your problem. It is related to Siemens PBX. With this topology, Asterisk is acting as a PSTN Central Office (a Public Central). What you asking is something like this: Asterisk acting as Central Office - HiPath - Public Central Office That is: the SIP devices connected to the Asterisk are not HI-Path's extensions! They seem external terminal/lines. So... You will have to enable, at HiPath, something called Transit or External traffic. In other words, it is a feature that you enable on HiPath allowing traffic between two trunks (the trunk connected to Asterisk and the trunk connected to the PSTN Central Office). Here we had to create a trunk access code. So, if a Asterisk user wants to call the outside number -1234, he/she will dial: 9 + -1234 Asterisk with then route this call to HiPath prefixing the trunk access code, for example, 88. So, asterisk will dial: 88 + -1234 Hope this helps, --hg - Original Message - From: [EMAIL PROTECTED] To: Pablo Allietti [EMAIL PROTECTED] Sent: Tuesday, October 25, 2005 11:52 AM Subject: Re: Siemens HI-path to ASTERISK Hi Pablo! I understood your problem. It is related to Siemens PBX. With this topology, Asterisk is acting as a PSTN Central Office (a Public Central). What you asking is something like this: Asterisk acting as Central Office - HiPath - Public Central Office That is: the SIP devices connected to the Asterisk are not HI-Path's extensions! They seem external terminal/lines. So... You will have to enable, at HiPath, something called Transit or External traffic. In other words, it is a feature that you enable on HiPath allowing traffic between two trunks (the trunk connected to Asterisk and the trunk connected to the PSTN Central Office). Here we had to create a trunk access code. So, if a Asterisk user wants to call the outside number -1234, he/she will dial: 9 + -1234 Asterisk with then route this call to HiPath prefixing the trunk access code, for example, 88. So, asterisk will dial: 88 + -1234 Hope this helps, Huelbe. - Original Message - From: Pablo Allietti [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, October 25, 2005 12:41 PM Subject: Re: Siemens HI-path to ASTERISK On Mon, Oct 24, 2005 at 06:42:02PM -0200, [EMAIL PROTECTED] wrote: Yes, for sure, it works. With TE110P from Digium using E1/ISDN/Pri signalling. By heart, I remember the following: 1. Configure Siemens E1 port as station and Asterisk as Pri_Net (or Central Office). 2. At Siemens, set the E1 port as S2 Point-to-Point net line without CRC4 or something like this. yep done. i only have a problem i can call any extension in the pbx but i can't take outside line with the 9 you can send to me the extensions.conf please please/ 3. At Asterisk, put these lines (/etc/zaptel.conf): span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 You have to study the rest of * conf file, but these ones are the important ones. Regards, --hg - Original Message - From: Pablo Allietti [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, October 24, 2005 6:55 PM Subject: [Asterisk-Users] Siemens HI-path to ASTERISK anybody can connect a Siemens HI-PATH to ASterisk via e1 ? i need your help please. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] iaxmodem
I succesfully compiled the spandsp-0.0.3pre4 version, but nothing changed no chance to compile the spandsp package boundled with iaxmodem Andrea [EMAIL PROTECTED] .it Sent by: To asterisk-users-bo Asterisk Users Mailing List - [EMAIL PROTECTED] Non-Commercial Discussion m.com asterisk-users@lists.digium.com cc Asterisk Users Mailing List - 25/10/2005 16.12 Non-Commercial Discussion asterisk-users@lists.digium.com, [EMAIL PROTECTED] Please respond to .com Asterisk Users Subject Mailing List - Re: [Asterisk-Users] iaxmodem Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Thanks for the answer. The problem went away starting faxgetty, I am sorry, I didn't carefully read the README Now I have another problem, which probably is exactly what Lee said, a spandsp version error. Now I am trying with the spandsp-0.0.3pre4 version. Andrea Lee Howard [EMAIL PROTECTED] van.com To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 25/10/2005 15.59 Re: [Asterisk-Users] iaxmodem Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com I'll see what I can do about improving the error messages involved with the write error, but that's not going to help your problem here. Until the IAXmodem documentation says otherwise, you *must* install and use the spandsp version that ships with IAXmodem. The only exception to this of which I know is that IAXmodem-0.0.5 will work with spandsp-0.0.3pre4 unaltered. The reason for this is because some of the IAXmodem development causes changes (fixes and enhancements) in the spandsp library, specifically the T.31 modem part. Eventually I expect these developments to slow down enough that IAXmodem will work with any version of spandsp after that point, but that point has not yet arrived. My guess is that the errors you're seeing are due to your not using the correct version of spandsp. As for the compilation error you're having with Autom4te (Can't locate object method path via package Autom4te::Request at /usr/bin/autom4te line 81.), I'm not sure what the right answer is. It's probably a dependency problem of some kind. So fix the spandsp build problem and I'll expect that your IAXmodem problem will go away... if not, then let me know. The IAXmodem forums or tracker is probably the better place for that, though. Lee. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
thank you for your answer I discovered that two lines, and now i removed the error in writing to the device (the device was not freed by faxgetty) now the problem is spandsp, If I call from a fax machine it rings 5-6 times and then it goes away (remote hungap) Unfortunately I am not able to compile spandsp, maybe it could be a perl-module problem ? (!?) No problem in compiling spandsp 0.0.3.pre4, but they are not OK for me Andrea Julian J. M. [EMAIL PROTECTED] omTo Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 25/10/2005 17.25 Re: [Asterisk-Users] iaxmodem Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com For hylafax to answer a call, you need to use faxgetty.. Add this 2 lines to your /etc/inittab and run init q to force a reload: IAX:2345:respawn:/usr/local/bin/iaxmodem ttyIAX modem:2345:respawn:/usr/sbin/faxgetty ttyIAX Change the paths according to your system. Julian J. M. On 10/25/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Probably you are right I installed hylafax and configured it to use iaxmodem, but I didn't start it Now I will research how to start hylafax, and I will try again Andrea ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
[EMAIL PROTECTED] wrote: now the problem is spandsp, If I call from a fax machine it rings 5-6 times and then it goes away (remote hungap) What your HylaFAX modem config file saying for RingsBeforeAnswer? Lee. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: changing protocols and transcoding
Hello all, forgive me if this is a simple question, but does bridging a SIP channel and an IAX channel that use the same codec (say, alaw) involve transcoding? i'm trying to figure out what kind of hardware i'll need, and i'm going to be using SIP endpoints and IAX trunking to move the audio along to another asterisk server(all with alaw), and i want to know if i'm going to need to figure transcoding into my hardware. I'm not familiar with the internals of IAX/SIP soagain forgive me if this is a dumb question. thanks, yair ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] connect 2 phones like in FOP
You could use a call file. This would be achieved like the following: exten = s,n,System(bash_file SIP/110 112) where bash_file is a script you make that drops a .call file into asterisk's outgoing directory. bash_file could contain something like this -- (from memory, research before you blame me): -- #!/bin/bash if [ $2 = ]; then echo Usage: $0 CALLER_CHANNEL CALLEE_EXTEN; else echo Exten: $2 Channel: $1 MaxRetries: 0 Priority: 1 Context: internalaugmented /tmp/somefile31329 mv /tmp/somefile31329 /var/spool/asterisk/outgoing done -- make sure to chmod +x bash_file after you create it If you can get this working, the example above should do something similar to making the phone at SIP/110 dial extension 112. Moj rulle mus wrote: Hello, Is it possible to connect 2 (SIP) phones via the dialplan. Sort of like dragging 2 phones to each other in Flash operator panel. The thing is I need an action in the dialplan that will connect 2 phones to each other as a reaction to an event without any intervention from one of the 2 phones. Regards, Mus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Siemens HI-path to ASTERISK
Hi Pablo, I really cannot forward the extension.conf due company rules. I am sorry. However, you are in the right path. If you can dial Hi-Path's extensions from Asterisk, you have 95% of the configuration done. All you need to do is: . enable on Hi-Path inter-trunk traffic. That is, traffic coming from a trunk has permission to sent through other trunk. . create an trunk access code so you can access the PSTN trunk from Asterisk's trunk . make Asterisk dial trunk-access-code + dialed destination. Please note here we tried to use the 9 access code (actually in Brazil we use widely 0 for outside call...) but we had some trouble, we had to create a double-digit trunk access code (it was 87, 88, 89, each one for a trunk from a different company). Something I remembered now: Siemens has something called block sent and non-block send configuration on ISDN trunk. It configures how digits show be treated (I think it is in block or one-by-one... sorry if I am saying non-senses here). You should try enable/disable this setting. Talk to your Siemens guy and ask him how to do this inter-trunk traffic permission. It is used a lot when you are interconnecting PABX from differentes brands (say Siemens + Alcatel). It also used when you have a trunk from a Telco company and wants to re-route the phone call to other destination using another Telco trunk. -hg - Original Message - From: Pablo Allietti [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 25, 2005 2:51 PM Subject: [Asterisk-Users] Re: Siemens HI-path to ASTERISK On Tue, Oct 25, 2005 at 12:31:41PM -0200, [EMAIL PROTECTED] wrote: Hi Pablo! ok. i do all the changes but now i have this error -- Channel 0/1, span 1 got hangup Oct 25 11:46:40 WARNING[3639]: app_dial.c:416 wait_for_answer: Unable to forward voice -- Hungup 'Zap/1-1' == No one is available to answer at this time -- Executing Playback(SIP/205-0014, invalid) in new stack -- Playing 'invalid' (language 'en') == Spawn extension (from-internal, 9122, 2) exited non-zero on 'SIP/205-0014' maybe is a extensions.conf ?? can you paste your extensions.conf here please? I understood your problem. It is related to Siemens PBX. With this topology, Asterisk is acting as a PSTN Central Office (a Public Central). What you asking is something like this: Asterisk acting as Central Office - HiPath - Public Central Office That is: the SIP devices connected to the Asterisk are not HI-Path's extensions! They seem external terminal/lines. So... You will have to enable, at HiPath, something called Transit or External traffic. In other words, it is a feature that you enable on HiPath allowing traffic between two trunks (the trunk connected to Asterisk and the trunk connected to the PSTN Central Office). Here we had to create a trunk access code. So, if a Asterisk user wants to call the outside number -1234, he/she will dial: 9 + -1234 Asterisk with then route this call to HiPath prefixing the trunk access code, for example, 88. So, asterisk will dial: 88 + -1234 Hope this helps, --hg - Original Message - From: [EMAIL PROTECTED] To: Pablo Allietti [EMAIL PROTECTED] Sent: Tuesday, October 25, 2005 11:52 AM Subject: Re: Siemens HI-path to ASTERISK Hi Pablo! I understood your problem. It is related to Siemens PBX. With this topology, Asterisk is acting as a PSTN Central Office (a Public Central). What you asking is something like this: Asterisk acting as Central Office - HiPath - Public Central Office That is: the SIP devices connected to the Asterisk are not HI-Path's extensions! They seem external terminal/lines. So... You will have to enable, at HiPath, something called Transit or External traffic. In other words, it is a feature that you enable on HiPath allowing traffic between two trunks (the trunk connected to Asterisk and the trunk connected to the PSTN Central Office). Here we had to create a trunk access code. So, if a Asterisk user wants to call the outside number -1234, he/she will dial: 9 + -1234 Asterisk with then route this call to HiPath prefixing the trunk access code, for example, 88. So, asterisk will dial: 88 + -1234 Hope this helps, Huelbe. - Original Message - From: Pablo Allietti [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, October 25, 2005 12:41 PM Subject: Re: Siemens HI-path to ASTERISK On Mon, Oct 24, 2005 at 06:42:02PM -0200, [EMAIL PROTECTED] wrote: Yes, for sure, it works. With TE110P from Digium using E1/ISDN/Pri signalling. By heart, I remember the following: 1. Configure Siemens E1 port as station and Asterisk as Pri_Net (or Central Office). 2. At Siemens, set the E1 port as S2 Point-to-Point net line without CRC4 or something like this. yep done. i only have a problem i can call any extension in the pbx but i can't take outside line with the 9 you can send to me
[Asterisk-Users] ECT - Specifying the transfer destination.
Hi
I have just a quick question on the README for the chan-capi-cm-0.6
relating to ECT.
In the first example case -
i.e.
exten = s,1,capicommand(ect|${MYHOLDVAR})
how is the destination number specified ? Is it implied somewhere ?
snippet from README ..
ECT:
Explicit call transfer of the call on hold (must put call on hold
first!)
Example:
exten = s,1,capicommand(ect|${MYHOLDVAR})
or
[macro-capiect]
exten = s,1,capicommand(ect)
[default]
exten = s,1,capicommand(hold)
exten = s,2,Wait(1)
exten = s,3,Dial(CAPI/contr1/1234,60,M(capiect))
.
thanks for any help,
best regards,
John.
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[Asterisk-Users] Echo cancel and fax
I have read that the digium t1 cards disable echo automatically if fax is detected, but I am assuming that this is hardware EC. I have 2 TE110P cards that, I believe, do not have HW EC. So, if I am using SW EC, does the EC still get cancelled on a fax call? If not, is there a way to control this. Current setup is PSTN---PRI---TE110P---Asterisk---TE110P---em_w T1---Panasonic DBS 576---Analog fax. We were using a PRI between the Asterisk and Panasonic, but the Panasonic's PRI card died. This had CID, so technically I could use that info before. But, now I am using a T1 card in the Panasonic, so I do not have CID and can not make outbound rules from that info. A replacement PRI card is $1900 and because Asterisk is to replace the Panasonic, I do not want to invest in it. Please advise. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE:[Asterisk-Users] H323 REGISTRATION PROBLEM: Gatekeeper '[EMAIL PROTECTED] ' found but failed to register
Hi all First of all excuse me if i make such a big post, hope also to write in the right place. I need to connect my linux/asterisk (10.0.0.252) box to a Nortel PBX (192.168.1.10) with h323 I'd like to allow some phones to register via sip to asterisk and with these to the Nortel PBX wich gives me the connections to the outside world (phone) after downloading and compiling the latest asterisk source from cvs OpenH323 v1.15.6, PWlib v1.8.7 (Mimas version from Voxgratia) and oh323-0.7.3 from http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.7.3.tar.gz starting asterisk i get snip Hi, I had the same problem in the same configuration. Asterisk finds the gatekeeper but it uses the wrong interface when it it should register. the problem is in the Mimas-patch2 release. change your pwlib to v1_9_1 and openh323 to version v1_17_2 then your registration works (again). Freddi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AudioCodes - TP260
We use tp-260 boards for ss7/sigtran.. they seem to behave similarly to mp2000 or tp1610 series boards which we have used with both mgcp and sip protocols.. their stuff seems to work rather well .. at least for us but YMMV. Chard Johnston wrote: Hi All, Does anyone have any experience with using Asterisk with AudioCodes TP260 SIP board? If yes, please let me know if you have had any problems. Regards, Chard Johnston ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancel and fax
On Tuesday 25 October 2005 13:07, Steven wrote: I have read that the digium t1 cards disable echo automatically if fax is detected, but I am assuming that this is hardware EC. You assume incorrectly. The zaptel software echo canceller also is disabled upon fax tone detection. This is on all zaptel products, from the lowly X100P to the upcoming DS3000P. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SER for dummies ?
Hi Have you got SER up and running If so then get asterisk up and running Then make sure ser can route to asterisk , search in google for routing to voicemail from ser, lots of people do that Now the call will be in asterisk, you will need to allow ser to pass calls, and vice versa ser needs to be told that asterisk is friendly. This is of course not the best user guide, but there isnt really one I have seen. Get the first two points up and running, post again... Iqbal Ralf Mueller wrote: Hello, I've been using Asterisk for a while now. For a large project I think about using SER, too. But although I have studied the SER tutorial, I'm not quite sure, how Asterisk and SER work together, how Asterisk know about clients that are registered at the SER and so on. Can anyone of you recommend a document or tutorial that explains this stuff for a dummy like me ? Thanks in advance. Ralf ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AudioCodes - TP260
Hi Matt, Thanks for the feedback. Regards, Chard. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Tuesday, October 25, 2005 1:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AudioCodes - TP260 We use tp-260 boards for ss7/sigtran.. they seem to behave similarly to mp2000 or tp1610 series boards which we have used with both mgcp and sip protocols.. their stuff seems to work rather well .. at least for us but YMMV. Chard Johnston wrote: Hi All, Does anyone have any experience with using Asterisk with AudioCodes TP260 SIP board? If yes, please let me know if you have had any problems. Regards, Chard Johnston ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax problem with zap trunk...
Someone has any idea about this issue? Thank you very much. Alexandre Leclerc a écrit : Hi all, as it is obivious at the bottom of this screen dump, when I'm recieving a fax from PSTN in the PBX, it fails to send it to extension 100 which is a fx_oks on my digium card. But I can call succesfully the fax from another internal phone (polycom ip600). Here is my zap config (I use [EMAIL PROTECTED]): # Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 fxoks=1 fxsks=2 fxsks=3 fxsks=4 Thanks for any help. Screen dump: -- Zap/2-1 answered SIP/108-e62e -- Starting simple switch on 'Zap/3-1' -- Executing SetLanguage(Zap/3-1, frqc) in new stack -- Executing GotoIf(Zap/3-1, 0?from-pstn-afthours|s|1:) in new stack -- Executing GotoIfTime(Zap/3-1, 7:55-18:05|mon-fri|*|*?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(Zap/3-1, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(Zap/3-1, ) in new stack -- Executing Wait(Zap/3-1, 1) in new stack -- Executing SetVar(Zap/3-1, intype=aa_1) in new stack -- Executing Cut(Zap/3-1, intype=intype|-|1) in new stack -- Executing GotoIf(Zap/3-1, 0?7:9) in new stack -- Goto (from-pstn-reghours,s,9) -- Executing GotoIf(Zap/3-1, 0?10:12) in new stack -- Goto (from-pstn-reghours,s,12) -- Executing GotoIf(Zap/3-1, 0?13:15) in new stack -- Goto (from-pstn-reghours,s,15) -- Executing Goto(Zap/3-1, aa_1|s|1) in new stack -- Goto (aa_1,s,1) -- Executing GotoIf(Zap/3-1, 0?4) in new stack -- Executing Answer(Zap/3-1, ) in new stack -- Executing Wait(Zap/3-1, 1) in new stack -- Executing SetVar(Zap/3-1, DIR-CONTEXT=general) in new stack -- Executing DigitTimeout(Zap/3-1, 3) in new stack -- Set Digit Timeout to 3 -- Executing ResponseTimeout(Zap/3-1, 7) in new stack -- Set Response Timeout to 7 -- Executing BackGround(Zap/3-1, custom/aa_1) in new stack -- Playing 'custom/aa_1' (language 'frqc') -- Alexandre Leclerc ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Format of extensions.conf
On 10/25/05, Sergey Okhapkin [EMAIL PROTECTED] wrote: Taking in the account poor Asterisk documentation, it's a bug. The bug can be called as a feature, only when it is documented:-) Poor Asterisk documentation? Ouch. Have you checked out http://www.asteriskdocs.org? We've just recently released Asterisk: The Future of Telephony published by O'Reilly there. You're more than welcome to contribute and help to resolve the lack of documentation problem. In fact I invite anyone and everyone to help with Asterisk documentation. We can't do it alone! Honestly though, with how huge Asterisk is, and its relatively new age, I think we're well on our way to being one of the _most_ widely documented projects. There is now the developers documentation hosted at http://www.asterisk.org/doxygen which has had numerous improvements over the last few days by Russell Bryant and Olle E. Johansson. Of course, theres the Asterisk Documentation Project at http://www.asteriskdocs.org The wiki: http://www.voip-info.org Oh, and links from Digium: http://www.digium.com/index.php?menu=documentation And of course, your friendly doc and configs directly located within your Asterisk source. And if you're still not convinced that documentation is being updated, check out the asterisk-doc list hosted at http://lists.digium.com for the CVS entries of all documentation which is constantly being updated. -- Leif Madsen - http://www.leifmadsen.com http://www.asteriskdocs.org -- Co-Founder http://www.oreilly.com/catalog/asterisk -- Co-Author ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] variable `oh323_tech' has initializer but incomplete type
Hi to all, i am trying to complie the openh323 for Asterisk. I have installed everything needed but when i try to do a make to asterisk-oh323-0.7.3 i get the following message: variable `oh323_tech' has initializer but incomplete type Any ideas? Thank you, Budoka. _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Redundency
Ok, I tried something slightly different.
I modified the existing the udp.monitor (or was it the tcp.monitor) of mon
and basically sending a sniffed SIP Registration packet which I send to
the asterisk server. If I don't receive an answer within a set time. The
monitor sends an error.
It tells you if the server is at least answering SIP. Mind you I once had a
server freeze, but the monitoring kept getting an answer. So not 100%
fool-proof, but save my *** in the past :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Moffett
Sent: October 25, 2005 9:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Redundency
Benjamin Lawetz wrote:
Since I can't do that, what I've settled on is heartbeat + mon.
Heartbeat will monitor for a system level failure and switch to the
backup
machine if neccesary; and mon will watch the asterisk (or any
other) service and restart it and/or alert me if it fails.
What kind of monitor are you using to monitor asterisk?
Sorry for my slow response. My asterisk monitor right now is embarrassingly
simple. All it does is execute show uptime and look for output starting
with System, see below. Obviously the method has limitations. 1) It will
only really only tell me that the daemon is running, not that it's able to
carry any calls. 2) It only works on localhost.
Input on how to test a remote instance of asterisk would be welcome, as well
as a method of making a test call or reliably testing for the ability to
make calls. My impression is that this would require asterisk to have a
Dial command in the CLI, or a linux SIP client that I could execute from
the shell. I'm not aware of the existence of either.
Any other simple and reliable methods of testing asterisk's condition would
be welcome.
The alerts, by the way are pretty simple as well. See the excerpt from
mon.cf below. restartasterisk.alert does exactly what it says.
stopeverything.alert shuts down heartbeat, which will cause another node in
the cluster to take over...in fact that node will start mon, which will then
use the restartasterisk.alert to start up asterisk. Asterisk only starts on
the backup machine when the primary fails so that config changes replicated
from the primary will take effect. Total downtime should be 3min. Which
will let me hit 5-nine if it only happens once a year ;)
Config changes are replicated via rsync and ssh every few minutes.
Voicemails are also copied from primary to backup by rsync. One thing I
still need to do is make rsync stop attempting to replicate files when the
failover occurrs. That will probably just require another alert below the
stopeverything.alert.
The replication of couse means that this setup will not protect me from a
bad config change that breaks asterisk, as that change will be replicated
throughout the cluster. So all significant config changes should be tested
on a standalone box.
[EMAIL PROTECTED] mon]# cat /usr/lib/mon/mon.d/asterisk.monitor
#!/bin/sh
##can only check localhost. Always checks localhost regardless of input
SHOW_UPTIME=`/usr/sbin/asterisk -rx show uptime | /bin/cut -b 1-6`
if [ $SHOW_UPTIME == System ]; then
exit 0
else
echo localhost
exit 1
fi
From mon.cf:
watch asterisk
service asterisk
description asterisk pbx on localhost
interval 10s
monitor asterisk.monitor
period wd {Sun-Sat}
alert mail.alert [EMAIL PROTECTED]
alert restartasterisk.alert [EMAIL PROTECTED]
alertevery 30s
service asterisk-failover
description checking if we need to stop heartbeat
interval 10s
monitor asterisk.monitor
period wd {Sun-Sat}
alert stopeverything.alert [EMAIL PROTECTED]
alertafter 5 3m
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Re: [Asterisk-Users] iaxmodem
The comment below makes me wonder could ttyIAX be configured to answer mgetty? I have made the mgetty talk to ttyIAX however, as soon as a ring comes into th eextension , mgetty shuts down... so I cannot keep the signal up. I tried to use the pppd daemon directly with ttyIAX and it said that the link is in serial loopback disconnecting. Would using iaxModem be feasable for a pppd dialin, or how could I use mgetty with pppd to start it? thanks For hylafax to answer a call, you need to use faxgetty.. Add this 2 lines to your /etc/inittab and run init q to force a reload: ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
[EMAIL PROTECTED] wrote: The comment below makes me wonder could ttyIAX be configured to answer mgetty? Although I've not tried mgetty with IAXmodem, the intent was to make this possible (for faxing), yes. I have made the mgetty talk to ttyIAX however, as soon as a ring comes into th eextension , mgetty shuts down... so I cannot keep the signal up. What does the mgetty logging say about what it's doing? I tried to use the pppd daemon directly with ttyIAX and it said that the link is in serial loopback disconnecting. Would using iaxModem be feasable for a pppd dialin, or how could I use mgetty with pppd to start it? spandsp (which is used by IAXmodem) does not currently support data calls. Lee. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to configure the communication between two Asterisk servers
Hi All: I have special set up to be done. See anyone can help me some ideas. Two Asterisk servers, server A trunks to PSTN, server B works as call routing engine. All sip phones are registered in server B. I have scenario like following: 1. A call comes to server A, server A sends the call related information to server B, assume that uses fast AGI. 2. Server B receives the message from server A, and look up dial plan for call routing, 3. Serve B sends the extension number back to server A, 4. Server A routes the call to the assigned agent. How does server B receive the message from server A? Many thanks for your help. Tielin Xu CTI Analyst Nintendo of America ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
mgetty dump: 10/25 11:23:17 IAX tio_get_rs232_lines: TIOCMGET failed: Invalid argument 10/25 11:23:17 # data dev=ttyIAX, pid=15158, caller='none', conn='DIRECT', name='', cmd='/bin/login', user='Fedora Core release 3 (Heidelberg)' -- 10/25 11:23:36 IAX mgetty: experimental test release 1.1.31-Jul24 10/25 11:23:36 IAX check for lockfiles 10/25 11:23:36 IAX locking the line 10/25 11:23:36 IAX tio_get_rs232_lines: TIOCMGET failed: Invalid argument 10/25 11:23:36 IAX WARNING: DSR is off - modem turned off or bad cable? 10/25 11:23:36 IAX lowering DTR to reset Modem 10/25 11:23:36 IAX TIOCMBIC failed: Invalid argument 10/25 11:23:36 IAX clean_line: only 500 of 4095 bytes logged 10/25 11:23:37 IAX waiting... i have in my /etc/inittab: /sbin/mgetty ttyIAX -F -r /dev/ttyIAX the -F is for Fax only and the -r is do not send modem init Then on the iaxmodem output i get a bunch of: Timestamp: 12001ms SCall: 06850 DCall: 00012 [192.168.1.1:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: PONG Timestamp: 12001ms SCall: 00012 DCall: 06850 [192.168.1.1:4569] Unknown IE 046 : Present Unknown IE 047 : Present Unknown IE 048 : Present Unknown IE 049 : Present Unknown IE 050 : Present Unknown IE 051 : Present Tx-Frame Retry[-01] -- OSeqno: 005 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 12001ms SCall: 06850 DCall: 00012 [192.168.1.1:4569] Unable to pass the full buffer onto the device file. -1 bytes of 4 written.Unable to pass the full buffer onto the device file. -1 bytes of 2 written.Unable to pass the full buffer onto the device file. 12 bytes of 25 written.Unable to pass the full buffer onto the device file. -1 bytes of 12 written.Unable to pass the full buffer onto the device file. -1 bytes of 2 written.Unable to pass the full buffer onto the device file. -1 bytes of 4 written.Unable to pass the full buffer onto the device file. -1 bytes of 2 written.Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: HANGUP Anyways.. it's a nice idea... and if spandsp supported data...it would be terrific!!! [EMAIL PROTECTED] wrote: The comment below makes me wonder could ttyIAX be configured to answer mgetty? Although I've not tried mgetty with IAXmodem, the intent was to make this possible (for faxing), yes. I have made the mgetty talk to ttyIAX however, as soon as a ring comes into th eextension , mgetty shuts down... so I cannot keep the signal up. What does the mgetty logging say about what it's doing? I tried to use the pppd daemon directly with ttyIAX and it said that the link is in serial loopback disconnecting. Would using iaxModem be feasable for a pppd dialin, or how could I use mgetty with pppd to start it? spandsp (which is used by IAXmodem) does not currently support data calls. Lee. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PDA softphone....
I have downloaded SJPhone - and well.. it does connect to my system, however popping audio is heard when i dial my music on hold extension... the quality is really really bad.. i have a WLAN-SDIO utility, the signal strength is at 11mb/s. however. is that sufficient? The codecs for sjphone are fixed at 64000.. i could not change those values. has anyone had successful attempts with something better? Thanks... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to configure the communication between two Asterisk servers
On Tue, 2005-10-25 at 11:15 -0700, Tielin Xu wrote: How does server B receive the message from server A? Many thanks for your help. Nintendo eh? The Redmond office? Thats near where I live. So let me make sure I understand the problem. Server A needs to get information from Server B about where to send the call to, which will most likely be somewhere from Server B, since all SIP phones go to server B? Why not use switch? We do something like that. We have 'Pandora' which is at a remote location connected to PSTN. We have 'Asterisk' which is local and all sip phones are connected to. 'Asterisk' has a context in dialplan that lists all the sip extensions and how to dial them and whatnot. 'Pandora' has a line within the context of the incomign PSTN calls that says: switch = IAX2/Asterisk/sipphones thats it! Basically it 'includes' the sipphones context on Asterisk into the call plan for Pandora. Works great. Does this help you? -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to configure the communication between two Asterisk servers
Tielin Xu wrote: Hi All: I have special set up to be done. See anyone can help me some ideas. Two Asterisk servers, server A trunks to PSTN, server B works as call routing engine. All sip phones are registered in server B. I have scenario like following: 1. A call comes to server A, server A sends the call related information to server B, assume that uses fast AGI. 2. Server B receives the message from server A, and look up dial plan for call routing, 3. Serve B sends the extension number back to server A, 4. Server A routes the call to the assigned agent. How does server B receive the message from server A? Many thanks for your help. Tielin Xu CTI Analyst Nintendo of America ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You can trunk your two servers through IAX or similar but I sense you are looking for something that goes beyond that though it's not too easy to discern from your messageWhy not have server B route the calls to the SIP agents registered on the same server B? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Redundency
A SIP registration as a monitor is not a bad idea at all. The
registration process is not too terribly complex, and I think I could
write a perl script that could attempt registration when supplied with a
host, username, and password.
No promises, but if I can put something together I'll post it.
Still, any ideas from anybody on how to make an automatic test call or
to simulate a call somehow would be appreciated.
Benjamin Lawetz wrote:
Ok, I tried something slightly different.
I modified the existing the udp.monitor (or was it the tcp.monitor) of mon
and basically sending a sniffed SIP Registration packet which I send to
the asterisk server. If I don't receive an answer within a set time. The
monitor sends an error.
It tells you if the server is at least answering SIP. Mind you I once had a
server freeze, but the monitoring kept getting an answer. So not 100%
fool-proof, but save my *** in the past :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Moffett
Sent: October 25, 2005 9:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Redundency
Benjamin Lawetz wrote:
Since I can't do that, what I've settled on is heartbeat + mon.
Heartbeat will monitor for a system level failure and switch to the
backup
machine if neccesary; and mon will watch the asterisk (or any
other) service and restart it and/or alert me if it fails.
What kind of monitor are you using to monitor asterisk?
Sorry for my slow response. My asterisk monitor right now is embarrassingly
simple. All it does is execute show uptime and look for output starting
with System, see below. Obviously the method has limitations. 1) It will
only really only tell me that the daemon is running, not that it's able to
carry any calls. 2) It only works on localhost.
Input on how to test a remote instance of asterisk would be welcome, as well
as a method of making a test call or reliably testing for the ability to
make calls. My impression is that this would require asterisk to have a
Dial command in the CLI, or a linux SIP client that I could execute from
the shell. I'm not aware of the existence of either.
Any other simple and reliable methods of testing asterisk's condition would
be welcome.
The alerts, by the way are pretty simple as well. See the excerpt from
mon.cf below. restartasterisk.alert does exactly what it says.
stopeverything.alert shuts down heartbeat, which will cause another node in
the cluster to take over...in fact that node will start mon, which will then
use the restartasterisk.alert to start up asterisk. Asterisk only starts on
the backup machine when the primary fails so that config changes replicated
from the primary will take effect. Total downtime should be 3min. Which
will let me hit 5-nine if it only happens once a year ;)
Config changes are replicated via rsync and ssh every few minutes.
Voicemails are also copied from primary to backup by rsync. One thing I
still need to do is make rsync stop attempting to replicate files when the
failover occurrs. That will probably just require another alert below the
stopeverything.alert.
The replication of couse means that this setup will not protect me from a
bad config change that breaks asterisk, as that change will be replicated
throughout the cluster. So all significant config changes should be tested
on a standalone box.
[EMAIL PROTECTED] mon]# cat /usr/lib/mon/mon.d/asterisk.monitor
#!/bin/sh
##can only check localhost. Always checks localhost regardless of input
SHOW_UPTIME=`/usr/sbin/asterisk -rx show uptime | /bin/cut -b 1-6`
if [ $SHOW_UPTIME == System ]; then
exit 0
else
echo localhost
exit 1
fi
From mon.cf:
watch asterisk
service asterisk
description asterisk pbx on localhost
interval 10s
monitor asterisk.monitor
period wd {Sun-Sat}
alert mail.alert [EMAIL PROTECTED]
alert restartasterisk.alert [EMAIL PROTECTED]
alertevery 30s
service asterisk-failover
description checking if we need to stop heartbeat
interval 10s
monitor asterisk.monitor
period wd {Sun-Sat}
alert stopeverything.alert [EMAIL PROTECTED]
alertafter 5 3m
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[Asterisk-Users] Re: Echo cancel and fax
Great. All of the references I read mentioned the card specifically, not zaptel or asterisk. Thanks for the info. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Andrew Kohlsmith [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] On Tuesday 25 October 2005 13:07, Steven wrote: I have read that the digium t1 cards disable echo automatically if fax is detected, but I am assuming that this is hardware EC. You assume incorrectly. The zaptel software echo canceller also is disabled upon fax tone detection. This is on all zaptel products, from the lowly X100P to the upcoming DS3000P. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PDA softphone....
As a secondary point, I'm looking at buying a Imate Jas Jar running windows mobile 5.0 to replace my treo. Does anyone have any thoughts on Windows mobile 5.0 specific softphones. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 25 October 2005 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] PDA softphone I have downloaded SJPhone - and well.. it does connect to my system, however popping audio is heard when i dial my music on hold extension... the quality is really really bad.. i have a WLAN-SDIO utility, the signal strength is at 11mb/s. however. is that sufficient? The codecs for sjphone are fixed at 64000.. i could not change those values. has anyone had successful attempts with something better? Thanks... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Redundency
Well what I was thinking of doing in the future was to have a cron job drop
a call file that would call another asterisk server that would auto-answer
and either generate some kind of network answer to MON or connect another
call to the first asterisk. Allows you to test your PRIs at a certain cost
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Moffett
Sent: October 25, 2005 2:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Redundency
A SIP registration as a monitor is not a bad idea at all. The registration
process is not too terribly complex, and I think I could write a perl script
that could attempt registration when supplied with a host, username, and
password.
No promises, but if I can put something together I'll post it.
Still, any ideas from anybody on how to make an automatic test call or to
simulate a call somehow would be appreciated.
Benjamin Lawetz wrote:
Ok, I tried something slightly different.
I modified the existing the udp.monitor (or was it the tcp.monitor) of mon
and basically sending a sniffed SIP Registration packet which I send to
the asterisk server. If I don't receive an answer within a set time. The
monitor sends an error.
It tells you if the server is at least answering SIP. Mind you I once had a
server freeze, but the monitoring kept getting an answer. So not 100%
fool-proof, but save my *** in the past :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Moffett
Sent: October 25, 2005 9:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Redundency
Benjamin Lawetz wrote:
Since I can't do that, what I've settled on is heartbeat + mon.
Heartbeat will monitor for a system level failure and switch to the
backup
machine if neccesary; and mon will watch the asterisk (or any
other) service and restart it and/or alert me if it fails.
What kind of monitor are you using to monitor asterisk?
Sorry for my slow response. My asterisk monitor right now is
embarrassingly
simple. All it does is execute show uptime and look for output starting
with System, see below. Obviously the method has limitations. 1) It
will
only really only tell me that the daemon is running, not that it's able to
carry any calls. 2) It only works on localhost.
Input on how to test a remote instance of asterisk would be welcome, as
well
as a method of making a test call or reliably testing for the ability to
make calls. My impression is that this would require asterisk to have a
Dial command in the CLI, or a linux SIP client that I could execute from
the shell. I'm not aware of the existence of either.
Any other simple and reliable methods of testing asterisk's condition would
be welcome.
The alerts, by the way are pretty simple as well. See the excerpt from
mon.cf below. restartasterisk.alert does exactly what it says.
stopeverything.alert shuts down heartbeat, which will cause another node in
the cluster to take over...in fact that node will start mon, which will
then
use the restartasterisk.alert to start up asterisk. Asterisk only starts
on
the backup machine when the primary fails so that config changes replicated
from the primary will take effect. Total downtime should be 3min. Which
will let me hit 5-nine if it only happens once a year ;)
Config changes are replicated via rsync and ssh every few minutes.
Voicemails are also copied from primary to backup by rsync. One thing I
still need to do is make rsync stop attempting to replicate files when the
failover occurrs. That will probably just require another alert below the
stopeverything.alert.
The replication of couse means that this setup will not protect me from a
bad config change that breaks asterisk, as that change will be replicated
throughout the cluster. So all significant config changes should be tested
on a standalone box.
[EMAIL PROTECTED] mon]# cat /usr/lib/mon/mon.d/asterisk.monitor
#!/bin/sh
##can only check localhost. Always checks localhost regardless of input
SHOW_UPTIME=`/usr/sbin/asterisk -rx show uptime | /bin/cut -b
1-6`
if [ $SHOW_UPTIME == System ]; then
exit 0
else
echo localhost
exit 1
fi
From mon.cf:
watch asterisk
service asterisk
description asterisk pbx on localhost
interval 10s
monitor asterisk.monitor
period wd {Sun-Sat}
alert mail.alert [EMAIL PROTECTED]
alert restartasterisk.alert [EMAIL PROTECTED]
alertevery 30s
service asterisk-failover
description checking if we need to stop heartbeat
interval 10s
monitor asterisk.monitor
period
Re: [Asterisk-Users] PDA softphone....
On Tue, October 25, 2005 20:27, [EMAIL PROTECTED] said: I have downloaded SJPhone - and well.. it does connect to my system, however popping audio is heard when i dial my music on hold extension... the quality is really really bad.. i have a WLAN-SDIO utility, the signal strength is at 11mb/s. however. is that sufficient? The codecs for sjphone are fixed at 64000.. i could not change those values. has anyone had successful attempts with something better? Thanks... Hello 'pbx' tongue mode=firmly in cheek Just wondering about two things... 1) What's your name? It's nicer to reply to someone by name... 2) What does this have to do with the thread in inter-asterisk communications? /tongue But seriously, we will need a bit more info, such as: What version? What platform? What type of network? (PS: Signal strength is something in dBi or %, 11mbps is the speed of the connection, but is it the actual speed, or just an indication of the type of network, ie 802.11b) What codec are you using? What config on the *? Have you looked at http://www.voip-info.org/wiki-Asterisk+phone+sjphone ? And please, next time start a new e-mail, and don't reply to an e-mail from the list. It screws up e-mail threading in proper mail-clients. (Your mail and my reply (and any others following) will be intermingled with those from the 'How to configure the communication between two Asterisk servers' thread) Even deleting all old text and the subject won't change that, as the threading is based on info in the headers or the e-mail you are replying to... If you cannot remember the list-address, make it in to a 'contact' for future use... ;-) Good luck! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PDA softphone....
On Tue, October 25, 2005 20:43, Dean Collins said: As a secondary point, I'm looking at buying a Imate Jas Jar running windows mobile 5.0 to replace my treo. Does anyone have any thoughts on Windows mobile 5.0 specific softphones. Cheers, Dean LOL! If you wait a bit longer you can buy a WinCE Treo! ;-) -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
