Re: [Asterisk-Users] OH323 doesnt hear ringing
On Mon, 26 Jan 2004, Aaron Martin wrote:
I have Asterisk running with a combination of SIP and H323 clients. I am using the
OH323 module instead of the H323 one.
When the SIP clients ring each other, they can hear a ringing noise in the ear peice
to let them know that the other parties phone is ringing. However, when the H323
client rings a SIP client, there is no ringing sound at all, although as soon as the
called party picks up the phone, everything works fine. This is the entry from my
extensions.conf:
exten = _7[5-9]X,1,Dial(SIP/${EXTEN},20,rt)
exten = _7[5-9]X,2,Playback(vm-nobodyavail)
exten = _7[5-9]X,3,Hangup
I assume that because I havr the 'r' in the dial string, the calling party should
hear a ringing noice. Any ideas?
I had the same problem, depending on which kind of channel I was calling.
Adding an explicit ring like this helped:
exten = _X,1,Ringing
exten = _X,2,Dial(SIP/${EXTEN},20,t)
exten = _X,3,Playback(...)
...
Cheers,
Siggi
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Re: [Asterisk-Users] I got it (was: Cisco 7940 with asterisk)
On Thu, 22 Jan 2004, Jeff Gustafson wrote: [...] no, its not necessary required. in this case, check that the contents of OS79xx.TXT if they match with your current version. I didn't have that file because I thought it would make things worse. :) I took the number from Settings - Model Information - App Load ID and put it in the OS79XX.TXT file. This doesn't seem to satisfy the phone. It still tries to pick up that .bin file. Now at least it's trying for the .bin file I specified. If I have the SEP*.cnf.xml file available, the phone just says Opening 192.168.5.254 then Opening 192.168.5.132 over and over again. The .132 is the right address for Asterisk. If I *remove* the SEP file, then the phone lets me dial, but doesn't let me get very far because it complains it can't see the .bin file. Thank you for your patience so far. I hope this process is helping others out there with cisco phones. Oh dear! I see all my newbie errors repeated here... So just to be safe: Remove _any_ files you might have on your TFTP server. The phone will try to fetch them, but boot fine even without them, provided that the configuration is reset, and there is an appropriate DHCP server. Once you have it working like that, you may think about TFTP config files again. In particular, these files will likely make things worse, if present: SMLDefault.cnf.xml SEPMACaddress.cnf.xml OS79XX.TXT Hope that finally helps... Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Grandstream 101
On Thu, 22 Jan 2004, dkwok wrote: Just got GS 101 phone and plugged into the network. Got ip setup however, the following problems arise: 1. when dialing an extension, I cannot further send any key tone to Asterisk. 2. there is no sound coming from the other end. [gs] canreinvite=no dtmfmode=info To solve 1., use dtmfmode = rfc2833 or just leave it empty. In the GS101 setting rtp port = 5004 sip port = 5060 dtmf = sip info using via RTP (RFC2833) here works fine for me. codec = pcmu codec = pcma Any pointer of a sample of config file would be most appreciate. WRT the codecs, Setting all 6 choices in the grandstream web interface has helped for me, most of the time. One phone required several reset cycles before it would accept new settings, though. Another one only accepted the new settings after unplugging/replugging the power supply. This one also lost its settings during another power supply. I guess these phones are just a bit flakey WRT their settings... HTH, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] I got it (was: Cisco 7940 with asterisk)
On Thu, 22 Jan 2004, Jeff Gustafson wrote: Sure: Remove the SIPDefault.cnf.xml file from your TFTP server! It's not required, and unless you understand what it's good for, it will keep your phone rebooting in an infinite loop. Well, it's XMLDefault.cnf.xml on my box. It doesn't have the SIP firmware load. Sorry, XMLDefault.cnf.xml is what I meant to say, of course... If your phone has some stale CallManager settings that are preventing you from using it, just follow these steps: 1) make sure you have a DHCP server on your network that will assign the phone an IP address, network mask, DNS server and default router It should also assign the address off your _Asterisk_ server as TFTP server (DHCP: next-server) That kinda helped. Now it's pointing right at Asterisk. The problem is still that I have better luck with the old chan_skinny code over the chan_sccp. chan_sccp still puts the phone in a config verification error loop. Did you remove the XMLDefault.cnf.xml? If so, there's still a load version conflict (although I've never seen a 7960 or 7940 care about the version communicated through SCCP): On the phone, press Settings, then 4 for load information. watch out for the App-Load-ID. On my 7940, this is P00305000300. Yours is most likely a smaller number... If you have a CVS version of chan_sccp (either Jan's CVS or Theo's), just add this exact string to your /etc/asterisk/sccp.conf in the device definition for your phone, ie. something like: [SEP00010203040506] imgversion = P00305000300 type = 7940 context = default autologin = yourline If you still have the 0.2 release of chan_sccp, I'd recomment you to upgrade, although you could change the value in sccp_helper.c, line 66 and recompile. If you have any further trouble, make sure you mention the App-Load-Id, because it seems to be an unusual one ;-) Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] I got it (was: Cisco 7940 with asterisk)
Hi Jeff, On Wed, 21 Jan 2004, Jeff Gustafson wrote: Kewl, I was apparently trying to use older chan_sccp code which didn't work. Okay... just tried your new code. The phones keep resetting: To be fair, one should mention that chan_sccp is actually hosted at http://www.zozo.org.uk/pages.shtml?page=sccp Unfortunately, Theo's computer is currently dead, so we've setup a copy of chan_sccp CVS in order to try and get the 7920 WLAN phones running. So the version from http://www.lambda-solutions.de/7920/ is in quite a state of flux, but does work somewhat with the 7920... Error Verifying Config Info then Registering [...] Any ideas? Sure: Remove the SIPDefault.cnf.xml file from your TFTP server! It's not required, and unless you understand what it's good for, it will keep your phone rebooting in an infinite loop. If your phone has some stale CallManager settings that are preventing you from using it, just follow these steps: 1) make sure you have a DHCP server on your network that will assign the phone an IP address, network mask, DNS server and default router It should also assign the address off your _Asterisk_ server as TFTP server (DHCP: next-server) 1a) If you cannot setup the DHCP to provide that TFTP server, make sure you have a host (IN A or CNAME) named ciscocm1 in your local DNS zone thet points to your Asterisk machine. 2) erase the stale configuration from your 7940 phone: - unplug ethernet (but keep power plugged) If you have PoE: you'll have to do the remaining steps quickly, or else the phone will interrupt you by rebooting - Press settings - type on the keypad: * - * - # - 3 (The **# part will unlock the settings, unless they are already unlocked. You can verify the little lock symbol in the upper right corner of the settings menu: there should be a tiny hole in the bail) - Select option Erase Configuration by scrolling down and pressing the Yes soft button when you reached it (My 7940 has it at position 33, so you can type 33 on the keypad to get there quickly, YMMV) - press the Save softkey and then back out of the setup. = the phone should reboot, get settings from DHCP and register to your asterisk, unless you have some wrong config files on the TFTP server, of course... HTH, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP Phones - Power over ethernet?
On Thu, 15 Jan 2004, Peter Pauly wrote: Are there any cheap SIP phones (like the Grandstream for example) that support power over ethernet? SNOM-105, SNOM-200, and all Cisco phones should support PoE. What is necessary to support SIP phones in a Cisco Call Manager environment? easiest solution: wait ~1 year until Cisco come's out with a SIP based CallManager on Linux. quick solution: Use Asterisk with chan_oh323 as an H.323 to SIP converter, configure it as an H.323 Gateway in CCM and register your SIP phones to *. Make sure to restrict codec selection on the Asterisk side: Translations, even from A-Law to Mu-Law, are likely to produce segfaults with this setup. Apart from that, it should work fine. Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Again: 7920 Cisco IP Phone Skinny SIP
On Wed, 14 Jan 2004, Jan Czmok wrote: It looks like chan_sccp is doing something at this pont that upsets the 7920 so that it tries to fall back to SRST mode, before finally re-registering. Okay, might be a reason. but what i saw on the display was: - Registering to Callmanager - Registered to Asterisk PBX - Call ended. (without hitting any button). Good point. When registering with CCM, it does send the SCCP messages for going off-hook and back on-hook again, but it does not display the prompts. However, that's just cosmetics, I guess (CCM doesn't have anything like the Registered to ... status message, and at the end of a call it always displays Your Options [...] Possible reason, looks like that, however i am surprised why the 7920 reboots instead of just looking for another callmanager. It does look for an SRST capable router first, then for up to 4 more CallManagers and only then reboots. (At least mine...) [...] There should also be a digitally signed version of that file (cmterm_7920.*.sbn), which the phone probably requires. nope. no sbn. according to my cisco source the file is not signed. Funny, that would be the first phone with unsigned firmware. But I'll double-check after the next firmware update. Some additional info: - The 7920 is requesting cmterm_7920.3.3-01-02-021^J.bin (so with an Ctrl-J in it), so you have to rename the file. You'd better remove the trailing Ctrl-J from OS7920.TXT, then (or stop using editors that insist on adding one). i did, i use vi as editor. i was also surprised, but this is coming from the phone ( i did extensive tethereal and tcpdump watching) :-) Umm, vi is notoriously adding Linefeeds at the end of file. Try echo -n cmterm_7920.3.3-01-02-021 /tftpboot/OS7920.TXT or use a decent editor! [...] I tried to use some 7960 images, but did not succeed :-( Of course not, it's totally different hardware. is it ? the cmterm image is nearly exactly 2 times the 7960 phone, so i suspect one lower part of the image for the new functions and the rest for the normal 79xx image. nah! Never. The 7920 doesn't have any support for XML-Services, for example. And I doubt that the Wireless hardware is in any way similar to the Ethernet switch in the 7960. [...] How can i help ? Just sit back isnt appropate for me :-) *shrug* Well, you could try to sniff the phone registering with CallManager (Express) and Asterisk during registration with chan_sccp. Maybe you'll see the difference that we're unable to spot. Meanwhile, you can register your 7920 with CallManager Express and connect that to asterisk via chan_oh323. (Note: chan_h323 will most likely not work, at least if you need two-way audio ;) Hmm. Might be one way to use CME. Will see.. Oh, wait! CME should come with a SIP capable router. Prolly better to use SIP, then. Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Again: 7920 Cisco IP Phone Skinny SIP
Hi Jan, first of all: please don't cross-post! On Tue, 13 Jan 2004, Jan Czmok wrote: [...] SKINNY OffHookMessage SKINNY SetSpeakerModeMessage SKINNY OnHookMessage SKINNY DisplayPromptStatusMessage SKINNY DisplayPromptStatusMessage SKINNY DisplayPromptStatusMessage It looks like chan_sccp is doing something at this pont that upsets the 7920 so that it tries to fall back to SRST mode, before finally re-registering. That re-registration is rejected by chan_sccp, though, as the old connection is not closed, yet. So the 7920 gives up and tries to find another CallManager. Right now, Theo as well as Martin Bene are looking at the packet traces, so I'm sure the issue will be located and fixed soon. But if you look at the Support of the 7920 in Callmanager Express, you get a file named cmterm_7920.3.3-01-02-021.bin so i was investigating further. so i wrote cmterm_7920.3.3-01-02-021 in OS7920.TXT and suddenly the Cisco 7920 shows Upgrading Firmware :-) Unfortunately for some reason it did not accept the firmware, but it still tries to load it. There should also be a digitally signed version of that file (cmterm_7920.*.sbn), which the phone probably requires. Some additional info: - The 7920 is requesting cmterm_7920.3.3-01-02-021^J.bin (so with an Ctrl-J in it), so you have to rename the file. You'd better remove the trailing Ctrl-J from OS7920.TXT, then (or stop using editors that insist on adding one). I also got the information from documents that the 7920 is running in 7960 emulation mode, so draw your own conclusions in regards of SIP possiblity :-) Nope, there is a statement from Cisco that SIP support for the 7920 is not planned, ATM. 7960 emulation mode refers only to being compatible with a 7960, as long as you do _not_ try to upload any firmware. (ie. Skinny-wise) However, that compatibility is not quite 100%... I tried to use some 7960 images, but did not succeed :-( Of course not, it's totally different hardware. Would appreciate some help in this issue :-) Just sit back and wait! Meanwhile, you can register your 7920 with CallManager Express and connect that to asterisk via chan_oh323. (Note: chan_h323 will most likely not work, at least if you need two-way audio ;) Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 7910 phone
On Tue, 13 Jan 2004, Ray Burkholder wrote: Will cisco 7910 ip phone compatible with Asterisk? I know [...] Cisco's site shows SIP drivers for 7960, 7940, 7912, 7905 only. If you want to run 7910 in Skinny mode, that may work. I'll leave that up to the chan_sccp and chan_skinny people. The 7910 has been reported to work with chan_sccp. However, if you're about to buy a phone, I'd strongly recommend you not to use the 7910, but either spend the few extra bucks for a 7960 or 7940 (only difference is the number of line/speeddial buttons) or take the 7905. While the 7905 doesn't have Speakerphone, its SIP image has by far the best user interface of all cisco SIP phones, plus it's nicely compact, and the display even has a higher resolution than that of the 7960... IIRC, it should also be the cheapest Cisco phone. Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Symbol NetVision Phone
On Tue, 13 Jan 2004, listas iPfone wrote: I received an unit of the Symbol NetVision Phone and i will test it with asterisk using H.323 or Skinny , somebody tested this phone with asterisk and can share experience? The phone itself is a bit flakey: turns off within 1 minute if it doesn't find it's access point (which is a PITA while configuring the access point's ESSID and WEP key), and for some reason, it stopped responding after a few calls, and the only way to get it to accept another call was power-cycling the phone. Maybe they have already fixed the firmware, though. Interoperability is fine though: it worked with any H.323 device, including both asterisk channel drivers. Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Again: 7920 Cisco IP Phone Skinny SIP
On Tue, 13 Jan 2004, Brian Buhrow wrote: Hello. The Cisco 7905 and 7920 phones are basically the same phone, with the 7920 having a built-in ethernet switch. Sip and Skinny images [...] Umm, the 7920 is Cisco's Wireless phone. It's definitely different from the 7905, and there's definitely no SIP image for the 7920 (whereas they have a really good one for the 7905). ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: [Asterisk-Dev] More Success on the Cisco 7920 and SCCP !!!!!
Hi Jan, the 7920 is on my todo list for quite a few days now, and I've had experience similar to yours... On Sun, 11 Jan 2004, Jan Czmok wrote: Latest status: chan_skinny does NOT work with 7920 chan_sccp does WORK with 7920 (!!) Yup. One should add that you'd better use the 0.2 release or a CVS snapshot. (Note for CVS: Zozo's server is a bit picky about its clients. You have to add the trailing / to your CVSROOT, exactly as documented. And even then, I was only able to make one checkout, whereas cvs update in that tree always failed. Some versions of CVS clients didn't work at all. One working version is the 1:1.12.2-2 Debian package) however: to remove coredumping the chan_sccp just comment out the MWI (messagewaitingindicator), then it compiles fine. Haven't had any trouble with that. (In fact, MWI works fine on a 7960 with chan_sccp here.) When do you get core dumps and do you have app_voicemail loaded/configured? Then change sccp_helper.c: return P0060302 instead of the old value. and voila: Phone is registering to Asterisk :-) But currently: -- After registering to Asterisk it received a off-hook message from the 7960 and then Call Ended on the Display (curious about that !!!). That seems to be normal for the 7920. I've sniffed the registration procedure with Cisco's newest 3.3(3) CallManager (+patches), and it's doing the same thing. Maybe that's some odd way of testing if the CallManager (CCM) really works... After that the phone reboots and the stuff repeats Same thing here. CCM does quite a few things in different order compared to chan_sccp, but apart from that, the registration procedure seems quite similar. I'm still looking into the detailed differences (which is a bit hard, as there doesn't seem to be any tool like diff for ethereal traces). Hope to find more answers soon, but it should lead people in the right direction. I hope so, too. And I'll let you know if there is any news. That said: The 7920 is definitely beta quality, at most: I have experienced dropped connections as well as the phone ignoring remote hangup, on more that 50% of my calls in a 100% Cisco environment, with a Cisco AiroNet access point within 3m, direct line-of-sight. So I guess it will take a year or two (just as with the 7960/7940) before Cisco gets the firmware in a state that can be considered production quality. Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Development Process comment and Email list suggestion
On Fri, 9 Jan 2004 [EMAIL PROTECTED] wrote: [...] Regarding the email list: A number of people have suggested creating more email lists. I think this is not a good idea because there will be even more cross posting than there is now between -dev and -users. That's a very valid point. [...] various modules of R/3 (ie: topic areas) were managed by using (and inforcing) topic keywords in the subject line. Enforcing the list guidelines was a 2 hour or so task each night for the moderators (we had several that rotated weekly shifts) but it made the list usage fairly easy. Subscribers could pick the topics of interest and then (assuming people followed the guidelines) their messages from the list were restricted to those topics. Dealing with chronic violators was always a judgement call by the moderator-on-duty: if a person was INTENTIONALLY not following guidelines then they were blocked. If they didn't understand the guideline then it was explained again to them and life went on. Generally, I like this idea. Note however, that there doesn't seem to be any dedicated moderator workforce for this list at all. (At least I've never seen any of the posts ever approved or rejected by a moderator that I've posted from the wrong address...) So for educating people we'd need some voluntary moderators, first. (No, I'm not volunteering.) [...] There would definately be a cost for the ListServe license (since there is a commercial profit from the list) but I think this software is best able to handle what we need to do. If Mark and Digium want to go in this direction I would be glad to coordinate the moderation. *hehe* Caught you here. Luckily, it doesn't take ListServe or any license. From the NEWS file[1] of MailMan 2.1: - Topic Filters o A new feature has been added called Topic Filters. A list administrator can create topics, which are essentially regular expression matches against Subject: and Keyword: headers (including such pseudo-headers if they appear in the first few lines of the body of a message). List members can then `subscribe' to various topics, which allows them to filter out any messages that don't match a topic, or to filter out any message that does match a topic. This can be useful for high volume lists where not everyone will be interested in every message. As that's exactly the feature you described, a simple upgrade of Digium's MailMan would cut it. As you're volunteering to organize the moderation, that should be a realistic plan, especially if you consider that the upgrade brings quite some bug fixes and improved performance due to better MTA integration... Cheers, Siggi [1] http://cvs.sourceforge.net/viewcvs.py/mailman/mailman/NEWS?rev=2.43view=markup ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asteriks as SIPH323 Proxy?
On Sat, 10 Jan 2004, Arnd Vehling wrote: is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC - SIP-Server - SIP/H323 Proxy - H323 Server - H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? Yes, Asterisk can do that fine. Unless you have an H.323 device with multiple IP addresses, like a CCM cluster with Skinny phones. In that case, you'll need chan_oh323. In any case, you'll have to install recent versions of openh323 (and pwlib) and build the H.323 channel driver (whichever you choose) manually. It won't work with all devices, though. (I have experienced some trouble when codecs don't match on H.323 and SIP sides, but that can usually be fixed by configuring allowed codecs explicitly, if all devices are under your control.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 79xx Ringtones
Hi, On Sun, 11 Jan 2004, Adthrawn wrote: I'm after two very specific ringtones for the 79xx's... A dog barking, and a horse either galloping or neighing. [...] I do recall, you had to set the sample length to a divisible, something like 800? And there was a maximum sample length too... Almost. The complete specs are on http://cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00800c2fa1.html#21431 That page also shows the complete procedure. If your phone doesn't use Skinny firmware, you might have to update RINGLIST.DAT instead of ringlist.xml (though I think the new SIP loads should also work with the xml file). HTH, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Multi-line help
On Mon, 5 Jan 2004, Philipp von Klitzing wrote: Am I correct by stating that as of now none of the VoIP protocols (SIP, MGCP, H.323, Skinny) supports such a silent ring feature? Would SIMPLE solve this? Naturally I wouldn't like those calls to show up in my list of unanswered calls. For the more expensive phones I guess you can assign a silent ring tone to specifc lines or caller IDs, but that'd only get me half the way. Well, Skinny, as a protocol, supports this. (You have to tell the phone explicitly, what state, caller ID and Name to show, if, and with which distinctive tone to ring, to flash the ring indicator, etc. You could even signal incoming calles as a status message and tell the phone to show a button template that contains some PickUp button. However, none of these features is supported in any Skinny Channel (and not in Cisco's CallManager, either) at this time. Many phone systems do support the notion of a silent ring only line, though. (CCM can do this with Skinny phones, most multiline SIP or H.323 phones have some silent ring option for particular lines, ...) SIMPLE would allow this too. But then, I do not know a single hardphone that supports SIMPLE. Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Testenvironment H.323 and SIP
On Mon, 29 Dec 2003, H S wrote: I am living oin Germany and having two ISDN BRI Lines available. Capi driver! I need a Sip Gateway and a H 323 Gateway. About H.323, there should be a full implementation of H.450. There is no such thing like a full implementation of H.450. Innovaphone's servers do come quite close, but then again, there's no phone that completely supports H.450. You should specify an exact set of call features that you really need. Which software is available that gives me a Sip and a H.323 Gateway to enter my PSTN with a BRI (digital line)? Asterisk sure can do that. But especially on H.323 many call features are not (or not properly) supported, yet. For SIP, it all depends on your terminal device: Some phones work fine (mostly Cisco), whereas others can make trouble with transfers, 3PTY or whatever call feature you can imagine... Is there anywhere I can find a board? (Forum) http://voip-info.org HTH, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interesting problem
On Thu, 18 Dec 2003, Christopher J. Wolff wrote: I have three cisco 7910 phones connected to * through skinny protocol. When one of the phones is called, and the phone is ringing, you can hear what's going on in the room even though the caller hasn't answered. It's crazy and very hard to ignore when someone is calling :) God forbid you should cough while the phone is ringing. If you call the Dial application without the r option, that should be fixed. The Skinny driver has a funny way of providing Ringback: It just switches audio through while ringing... If you still need ringback, try doing it explicitly like this: exten = 123,1,Ringing exten = 123,2,Dial(Skinny/[EMAIL PROTECTED]) ; do NOT add |r or ,r here!^ HTH, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7912 speed dials
On Sun, 21 Dec 2003, Ludovic Drolez wrote: We have Cisco 7912 phones, and the doc says that I can create up to four speed dial buttons on my phone using the Cisco CallManager. Does anyone knows which protocol is used to configure speed dials (Is it documented somewhere) ? Did someone tried to reverse engineer the protocol ? Yes, it's the Skinny Client Control Protocol, aka Skinny or SCCP. There are 2 Asterisk channel drivers supporting this protocol: - chan_skinny, which comes with asterisk This one is quite stable but supportsy nearly nothing except placing or receiving single calls. - chan_sccp, from http://theo.me.uk/pages.shtml?page=sccp This one may crash asterisk if some very odd skinny phone tries to register. (The 7912 may qualify as odd, as it's not officially supported right now.) However, it does support some nice features (soft keys, multiple calls don't crash *, ...) Some features like intercom (paging) or speed dials don't seem to be completely implemented, yet. But the author seems quite responsive to me and the code is easy to read/update. It would be cool, not having to pay $15000 just for configuring speed dials on those phones ;-D Sure, but some development work still has to be done, first. Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and cisco call manager via h.323
On Tue, 16 Dec 2003, Pavel Zheltouhov wrote: Does asterisk work with CCM as gateway ? When I trying call asterisk,I totally can't hear any sound. When call ohphone - works good. 10.0.1.219 is CCM, 10.0.1.207 asterisk. [...] Tested with latest cvs asterisk. Maybe asterisk h.323 channel driver not correctly parse h.323 messages. Yes, it's a known bug in chan_h323: Asterisk's RTP stack will blindly dump the RTP stream to CCM instead of the phone's IP address, which CCM correctly told us during H.245 setup. chan_oh323 from http://www.inaccessnetworks.com/projects/asterisk-oh323 will work fine. HTH, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Using asterisk as voicemail with SER
On Wed, 17 Dec 2003, Victor Medrano wrote: i did with cisco callmanager with smdi integration . and h323 . works very well . You got SMDI working with CCM? How? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debian Testing / 2.4.22 / zaptel problems.
On Thu, 4 Dec 2003, Andrew Gillham wrote:
Well as far as I can tell, the only version I have on the box is 2.4.22-1.
I certainly only have 'kernel-headers-2.4.22-1' installed and 'linux'
symlinked
to that directory in /usr/src.
Are you saying my /usr/include would be skewed? Since I thought that
was from
the libc6-dev, not really kernel related?
That's true, for anything _but_ compiling kernel modules.
You have to tweak the zaptel Makefile to point to your kernel-headers'
include directory. something like this should do:
KINCLUDES=/usr/src/kernel-headers-2.4.22/include
I had to add -I$(KINCLUDES) to KFLAGS, too, for some obscure reasons, but
maybe that's not necessary, any more.
I will try removing all of the -dev packages and re-installing them.
That won't help.
And I'd _strongly_ suggest you don't point /usr/include/{linux,asm} to
your current kernel instead of the headers that come with libc. I know
it's tempting, but it will buy you lots trouble if you decide to upgrade
libc...
Cheers,
Siggi
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Re: [Asterisk-Users] Configuring CISCO IP 7940 for *
On Tue, 2 Dec 2003, tony banks wrote: I have 1 IP 7940 with the following Firmware versions App Load ID: P00303011201 Boot Load ID: PCO303010001 Version 3.1(12.1) Sounds like a Skinny image to me (and an old one, around CCM 3.1, too.) You can either get a SIP image from cisco and flash your phone with that or try one of asterisk's Skinny channels: - chan_skinny comes with asterisk CVS, is quickly setup, quite stable and provides basically no features except placing and receiving simple calls - chan_sccp supports multiple lines, softkey templates, fastdial buttons, but it may segfault more easily and you'll have to download it from http://www.zozo.org.uk/pages.shtml?page=sccp Both are not really production quality right now, so you'd probably be better off flashing your phone with either MGCP or SIP Firmware. OTOH, your firmware is old enough to not require a signed firmware image. So you could theoretically port an IAX stack to the phone ;-) HTH, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mailing list configuration issues...
On Wed, 19 Nov 2003, Florian Overkamp wrote: [...] add a link to the FAQ on the bottom of every message, much like the subscribe/unsubscribe instructions. Then again, nobody seems to read those either :-P... Not sure about what mailing list footer you get, but I do only get this at the bottom of each mail: Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users No unsubscribe instructions in there, unless you know that this is mailman, where you unsubscribe using the listinfo form. That said, I'm sure proper unsubscribe instructions (To unsubscribe, go to http://lists.digium.com/mailman/listinfo/asterisk-users or send a message with just the word 'unsubscribe' to [EMAIL PROTECTED]) would help a great deal. While at it, the admin could also enable Mailman's Administrativa filter, which efficiently blocks unsubscribe messages to the list. That said, I find an FAQ quite a good idea. Maybe just as another page on the voip-info.org Wiki? Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Message lamp integration with legacy pbx during conversion
On Sun, 16 Nov 2003, Josh J. Zuerner wrote: [...] For example, if I currently dial 1000400 on my * SIP phone, the MW lamp on legacy X 400 is flipped on by the PBX. If I dial 1001400 on my * SIP phone, the MW lamp on legacy X 400 is flipped off. Does this dialing capability already exist? I would think that in any case where * would be co-implemented with another PBX, some mechanism for basic DTMF voicemail notification would be needed. Alternatively, support in voicemail.conf for a newvoicemail event command and a nonewvoicemail event command would be very helpful. Just have a look at the sample.call file that comes with asterisk. You'll only have to fill out the right telephone numbers and maybe have it connect to some dummy extension that just waits a second before hanging up (if your PBX needs more than just a ring). Then put a copy of that file in /var/spool/asterisk/outgoing and * will immediately place the call you described in that file. I appreciate any assistance and would also take feedback on paying for someone to contribute this functionality to *. We are really excited about using the platform in our office, and are doing so on a limited basis, but we need this support to move forward. If you feel the urge to spend money, just contact me off-list ;-) Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN (isdn4linux) DDI
On Tue, 11 Nov 2003, Matthew Enger wrote: I have it working on distinguising just the local numbers of our 4 B channels and the number assigned to the group. I have ordered an '100 in-dial range' here in Australia and should have it available to me by the end of next week, I can let you know how it goes. Cool. That would make 100 virtual voice modems with I4L ;-) I guess you'd rather use chan_capi if you're needing more than a handful of numbers! Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Polycom Soundpoint IP600
On Mon, 10 Nov 2003, Chad Cowan wrote: This Polycom phone seems to be one of the best on the market for sound quality and features. I have seen on the list that some people have gotten the IP 600 to work with Asterisk. Does anyone have the details of how to get this working i.e. XML phone config files, and any thing else I might need to know. Isn't this the original Cisco 7960? (ie. Cisco licensed the technology from Polycom and basically put it in a neatly designed new case with less hard keys) Or am I mixing things up with the IP 500?! Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Making a Skinny phone talk to Asterisk
On Sun, 2 Nov 2003, Florian Overkamp wrote: At 15:07 1-11-2003 -0600, you wrote: Last I checked skinny firmware would try to connect to a host that would resolve to CiscoCM1 Actually that is just a last-resort. Before that it will try and find the callmanager by looking for some special DHCP flag, and if that is not around it will try the setting for next-server (which is a DHCP option also). Some phones even try the DHCPserver itself if the above fails. Funny, mine always tries to connect to CiscoCM1 _first_, but maybe I just failed about the right DHCP setup... You can also run ith with totally static configuration though: Just clear all network settings, disable DHCP and configure a static IP address, Netmask Gateway via the phone's setup menu. It will then connect to whatever server you have told it to use as TFTP server on next boot, ie even if you don't have a TFTP server running, just set this to point to your Asterisk box, and the phone will register there (after failing to connect to CiscoCM1). If you _do_ have a TFTP server: even better. You can just put some XML files there that the phone will read its settings from. HTH, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Apple implementation
On Wed, 5 Nov 2003, Dave Weis wrote: On Wed, 5 Nov 2003, Charles Hatchette wrote: I am new to Asterisk and Digium card implementation issues. My VAR is strongly recommending using Apple hardware and Yellow Dog Linux for my telephony project, because of his familiarity with this OS. Is the PowerPC an appropriate and stable hardware platform for Digium/Asterisk development? I don't know if the drivers for the cards have been tested on ppc machines. * is running fine on an iBook G3/600 (Debian GNU/Linux) here. Hardware may be a problem, though. At least Digium's USB FXS adapter is working here, even though the driver used to have some intel only days. (ie. it would not compile on PPC, due to some hard-wired MMX stuff, but that was fixed in CVS after a few days.) I guess the other Digium hardware should work just as well. Many other vendors require binary-only (and therefore x86 only) firmware upload tools, though. HTH, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: Restoring Cisco 7960 to defaults
On Tue, 14 Oct 2003, Sales wrote: Can anyone point me to some online documentation showing how to reset a CP-7960 to factory default settings. I have some that are configured for Callmanager and I want to get them back to generic default config. Any info is appreciated. - unplug ethernet and power - plug in power _only_ (no net!) - press Settings, *, *, #, 3 - scroll down to the erase config option (no. 33 with my firmware) - select yes, then save - unplug/replug power again (now it's all set to default and you can use the same procedure to enter new settings...) Ton't forget to plug the Ethernet cable back in after configuration! Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 79xx XML carriage returns/line feeds
On Thu, 21 Aug 2003, Low, Adam wrote: I've been developing all sorts of applications for use on our 79xx handsets but am having great difficulty with formatting, I just can't seem to be able to produce a line feed between lines on the stuff actually displayed on the phone. Has anyone else has experience or success with this ? Which version of the phones are you using? SIP, MGCP or Skinny? And which XML dialect? (CMXML or BTXML?) Using CMXML on Skinny firmware, CR characters (ASCII 13, \r) are displayed as CR/LF. Do not use LF (ASCII 10, \n), as they are displayed as an odd graphic symbol. CMXML also doesn't know anything like br/ tags... HTH, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extension and phone management bestpractices??
On Wed, 13 Aug 2003, Devon Henderson wrote:
[...]
We have agents who work both from home and from the office.
Some agents are
always in the office, some are always at home, and some
alternate between
the two.
[...]
I guess my big question is: is it possible to have extensions mapped to
people, not to phones?
Sure, as mentioned by others, you'll just have to map user extensions to
phone lines when users are logged in.
I'd strongly recommend assigning numbers to to individual phones, too.
(Sometimes you want to call room No. 123, no matter who's logged in
there, eg if the PC in there suddenly stops responding and you want
somebody to look after that...) Those location based extensions would be
independent from the technology used, so you could exchange MGCP, SIP or
analog phones by just reassigning the phone number to a new channel.
That said, you can quite nicely map user extensions to location extensions
using *'s extension logic and the database application. Let's assume your
inhouse policy says:
Extensions 5XYZ refer to Phone No. Z in room No. Y on Floor X,
Extensions 6XXX refer to user no. XXX.
You would then start by conventionally mapping your phone lines to the
5XYZ extensions in extensions.conf (and the appropriate channel
configuration files).
Next step is adding your users to *'s database, assigning them a password,
eg:
CLI database add password 001 123
CLI database add password 002 456
(That assigns user 001 the password 123, whereas user 002 gets 456.)
Now you can add quite simple extension logic for all users in your
database, eg. like this (extensions.conf, again):
exten = _6XXX,1,DBGet(TARGET=location/${EXTEN:1})
;user location was found in DB, go there (assuming _5XXX in default)
exten = _6XXX,2,Goto(default,${TARGET},1)
;DBGet failed, so user is not logged in. = give VoiceMail
exten = _6XXX,102,Voicemail2(u${EXTEN})
You'll probably want to set outgoing caller ID to the user's extension
instead of the location-based one. That's easy, too:
Put all the phones into a special context that has this:
exten = s,1,DBGet(USER=UID/${EXTEN:1})
exten = s,2,SetCallerID(6${USER})
exten = s,3,Goto(default,s,1)
Finally, you'll need some way for users to register. That could be a web
form that checks the user-provided password (using DBget, maybe via
asterisk -rx database get password XXX), and puts the given location
into the database (using DBput, maybe via
asterisk -rx database put location XXX 5XYZ)
The same script must set the reverse mapping, too, using DBput or like:
asterisk -rx database put user 5XYZ XXX
You could also do that via extension logic, eg let people just dial
7XXX to log in, using something like this:
(note that I leave the password check as an exercise to the reader ;-)
; login user if they come from a location (5XXX) extension
exten = _7XXX/_5XXX,1,DBput(location/${EXTEN:1}=${CALLERIDNUM})
exten = _7XXX/_5XXX,2,DBput(user/${CALLERIDNUM}=${EXTEN:1})
exten = _7XXX/_5XXX,3,Playback(login-succcessfull)
exten = _7XXX/_5XXX,4,VoiceMailMain(s6${EXTEN:1})
And to logout, just have people dial 7000:
exten = 7000/_6XXX,1,DBget(LOCATION=location/${CALLERIDNUM:1})
exten = 7000/_6XXX,2,DBdel(user/${LOCATION})
exten = 7000/_6XXX,1,DBdel(location/${CALLERIDNUM:1})
exten = 7000/_6XXX,2,Playback(user)
exten = 7000/_6XXX,3,SayDigits(${CALLERIDNUM:1})
exten = 7000/_6XXX,4,Playback(logged-out)
Phew!
That has become more that I had expected. Anyhow: You should get the idea
that it's possible (and quite useful) to have _both_ locations and users
assigned to constant extensions using Asterisk, and you can do the login
via both a web form (CGI, PHP, whatever) or directly by having people dial
magic extensions.
Cheers,
Siggi
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Re: [Asterisk-Users] Asterisk as a stand alone voice mail server(fwd)
Hi again. On Mon, 11 Aug 2003, Rainer Jochem wrote: I've played around a little bit and discovered the following: with services_url: http://xxx.xxx.xxx.xxx/xmlservices/vm/index.php?user=1234amp;pin=1234; the phone tried to get GET /xmlservices/vm/index.php?user=1234?pin=1234name=... changing the services_url to: http://xxx.xxx.xxx.xxx/xmlservices/vm/index.phpamp;user=1234amp;pin=1234; the phone requests GET /xmlservices/vm/index.php?user=3841pin=3841name=... Whew! These phones seem to do some quite strange stuff... if I enter this URL in my Browser, it works. But the phone still says CMXML Error Okay, you already told me (in pm) that your phone does this whenever it gets any SoftKey definitions. I guess it's time for Cisco support to solve this mystery... So I guess there's something in the xml-files the phone doesn't understand. If you want to check more: I've finally put some test scripts on http://swt.uni-stuttgart.de/~langausd/asterisk Greetings from too sunny Saarbruecken Bah! It's at least that sunny here in Stuttgart. Plus we do not have any significant wind, being encircled by mountains. The City has had max. temperatures of 40°C during the past 4 days. *sweat* Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ip phones and intercom/paging
On Sat, 9 Aug 2003, John Todd wrote: The Ciscos should be able to do this via a XML Service using the RTPTx: and RTPRx: URI schemes. The tricky part is just pushing the RTPRx to the listening phone. XML service specifications are here: http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide09186a008018e234.html I don't think this is possible, from my understanding of the Cisco firmware using SIP images. While it may be possible to initiate receipt of an RTP stream from an XML command, how do you get the Cisco to execute the XML command to begin with, without having the user push keys to start up the process? Caution: to not believe the Cisco manuals are correct when they say that the SIP O/S on the phone runs a web server. They do not. Really?? Well, the Skinny version does run a web server. Maybe an upgrade of the SIP firmware comes with that feature? The only way that I have discovered the Cisco 79xx phones can download a URL from a remote server are via either user interaction (pressing a sequence of keys) or by putting together a really UGLY XML page that refreshes every few seconds. But even this suffers from the problem of: How does the first page get requested? The phones have an idle URL setting, which can be used exactly for that: polling an XML service every n seconds. However, I admit that's ugly. I'd be happy to be proven wrong in my assertion that intercom and paging is impossible with Cisco SIP phones. Please send code. :) Will do, as soon as I get time to play with this. That's way at the bottom of my todo list, though... Now, if all your comments are about SCCP phones, can you send along the SCCP driver for Asterisk so the rest of us can play? Well, sorry. For now, you'll have to use CCM and configure * as an H.323 gateway using chan_oh323. Writing a Skinny driver is definitely somethinh I will not do. I'd happily move away from that protocol, if they'd just let me... Maybe the MGCP firmware is an alternative? PS: If one could initiate XML requests remotely, perhaps a better way to do intercom/paging would be to use the Play command, and (if possible) use a pipe on the server side to connect to a live audio stream. (sigh) If only I could control the phones from the server... Nope, the play command (whatever you mean by that: there's playing a file via TFTP, and playing audio/basic via HTTP) won't work properly for that purpose: The phone would just download the whole stream before playing it (in both cases). That means you get an initial delay = length of stream, and the stream is limited by the phone's memory (to 5s according to the documentation, to 20s acording to my experiments...). Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a stand alone voice mail server
On Wed, 13 Aug 2003, James Sizemore wrote: Leave off the softkey xml tags, This should get you working You can telnet to the phone and type debug http and you will get better errors. While that seems to work on the 7960 SIP firmware, it's not quite satisfactory. Are softkeys officially unsupported by the SIP phones??! Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ip phones and intercom/paging
On Fri, 8 Aug 2003, cwitte wrote: There was a thread a few months ago that tossed around some ideas for using a cisco phone for intercom or paging. I don't have any ip phones, and wondered if anyone had any luck getting intercom or paging to work on the cisco units. The Ciscos should be able to do this via a XML Service using the RTPTx: and RTPRx: URI schemes. The tricky part is just pushing the RTPRx to the listening phone. XML service specifications are here: http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide09186a008018e234.html Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Semi-newbie question Softswitch and Asterisk- Is there a difference?
Hi Bruce, On Wed, 6 Aug 2003, Bruce Ferrell wrote: [...] Is there a difference between what asterisk is and a softswitch? Can someone explain it in small words and phrases for me? Sure, there is. First of all, * is much cheaper. But technically, * does much more than a soft switch: AIUI, a classicals soft switch _only_ serves the purpose of managing call setup for a group of terminal devices. * is more like a classical PBX: Terminal devices can actually connect to *. Asterisk can either simply transfer calls to the correct location (much like a soft switch), but it can also function as a (quite universal) telephony gateway (or proxy). Additionally, * provides advanced IVR scripting, Voicemail, call queueing, and with some tweaking: a text-to-speech gateway, or even voice recognition. I guess you get the picture, but let's try to list asterisk's features along with their classical VoIP pendant (AFAIK them) in brackets: - manage call setup between end devices and gateways (soft switch) - Convert VoIP calls to analog (or ISDN) lines (POTS or ISDN gateway) - Connect otherwise incompatible VoIP devices, eg SIP--H.323 (advanced soft switch) - Convert incompatible audio codecs, eg GSM to G.711 (?) - Provide outside access for an otherwise firewalled IP phone network (VoIP proxy) - Voice Mail (voice mail system) - IVR applications (IVR server) - Music on hold source (?) - ... (I'm quite sure I forgot something.) As you can see, I'm not quite a specialist in classical telephony systems (or maybe some of *'s functions just do not have a classical pendant), but I guess you got an idea. Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Wildcard x100p support Caller ID outsidethe US? (fwd)
On Fri, 8 Aug 2003, Dave Cotton wrote: The x100p does get the CID in France. It is now a question of how to break it down. I changed callerid.c line 278 to :- ast_log(LOG_NOTICE, Got this:- %s\n, cid-rawdata); and the result on August 8 at 10:06 from 0490233081 was:- File callerid.c, Line 278 (callerid_feed): Got this:- 080810060490233081 OK, Now what do I do? Well, I'd say: just strip the date and time off!. You made that call on 2003-08-08, 10:06h local time, didn't you? Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a stand alone voice mail server(fwd)
On Fri, 8 Aug 2003, Maik Schmitt wrote: [...] I just tried to use it with our 7960 (sip-version). I've set the services_url in SIPDefault.cnf to http://xxx.xxx.xxx.xxx/xmlservices/vm/index.phpamp;user=1234amp;pin=1234; It didn't work with ?user=...pin= cause the phone then tried to get index.php?user=1234\x9fpin=1234?name=SIP... Okay, I almost anticipated that. You're lucky, as my Skinny phones even have a tendency of crashing quite strangely (sometimes rebooting, sometimes just messing up the application flow..) on unescaped XML characters... Now I only get: --- CMXML Error XML Parse Error Hmmm. I never got one like that. (I haven't ever tried SIP firmware, though...) Which firmware version are you using? Have you ever had any XML services running on these phones? If so: Could you provide me with sample XML that works? I'll put some simple test scripts online later today, so maybe you want to try them... When I open the URL with my Browser I see the VoiceMail - INBOX and the source also looks OK: --- CiscoIPPhoneText TitleVoiceMail - INBOX/Title Prompt/Prompt Text^MEs sind keine Nachrichten vorhanden./Text SoftKeyItem NameBeenden/Name URLSoftKey:Exit/URL Position3/Position /SoftKeyItem SoftKeyItem NameOrdner.../Name URLhttp://xxx.xxx.xxx.xxx/xmlservices/vm/index.php?user=1234amp;pin=1234amp;folder=INBOXamp;do=chfolder/URL Position4/Position /SoftKeyItem /CiscoIPPhoneText --- Any Ideas? Not really. It does look perfectly okay to me. (it should say Eingang instead of INBOX, though, unless you changed the translation...) You could try leaving yourself voicemail, first. That should give you a CiscoIPPhoneIconMenu instead of the CiscoIPPhoneText. _maybe_ the phone is happier with that... Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a stand alone voice mail server
Hi again, this just popped into my eyes: On Fri, 8 Aug 2003, Maik Schmitt wrote: I just tried to use it with our 7960 (sip-version). I've set the services_url in SIPDefault.cnf to http://xxx.xxx.xxx.xxx/xmlservices/vm/index.phpamp;user=1234amp;pin=1234; you mean: http://xxx.xxx.xxx.xxx/xmlservices/vm/index.php?user=1234amp;pin=1234 Of course, don't you? (ie: keep the initial ? and only replace by amp;) Maybe that's already it. (With amp; instead of ?, Apache should give you a 404, wich makes the telephone produce some error, of course...) In case you get it working with the sercices URL, could you try the messages URL, too? At least with the Skinny firmware, the phone seems to add pieces of XML for a non-existing Unity Voicemail system... Good luck, Siggi PS: looks like the test scripts will only come tomorrow. *sigh* *sweat* ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a stand alone voice mail server
Hi again,
struggling with localization issues (so the script is not German only)
took me a week longer than expected. (Did anybody ever get PHP's gettext
extension working??)
But finally, I've wrapped something up:
On Thu, 24 Jul 2003, Dave Packham wrote:
I would like to see your code...
sounds great
Well, don't expect too much!
I've wrapped up an archive along with some elementary documentation. You
can download it from
http://swt.uni-stuttgart.de/~langausd/asterisk/
You will need something like the attached patch, so app_voicemail2 writes
files in a way that allows PHP to read them.
If I ever understand how to add config options to asterisk applications,
I'll do it properly...
Note that I don't even have a proper name for the project, yet.
(suggestions are welcome!)
Due to the early stage of the project, I haven't really shipped it with
any usable license. If anybody really wants to use this now: please
contact me off-list! (It will probably be something like BSD Artistic
license if it ever becomes a real project...)
Otherwise: feel free to test this stuff, and rip out what you want!
(The only exception is: cisco-xmlapi.inc.php Please don't distribute that
as-is. I'm currently revising the whole API...)
Cheers,
Siggi
Index: apps/app_voicemail2.c
===
RCS file: /usr/cvsroot/asterisk/apps/app_voicemail2.c,v
retrieving revision 1.35
diff -u -r1.35 app_voicemail2.c
--- apps/app_voicemail2.c 2 Aug 2003 21:10:06 - 1.35
+++ apps/app_voicemail2.c 8 Aug 2003 15:55:52 -
@@ -58,6 +58,23 @@
#define BASEMAXINLINE 256
#define BASELINELEN 72
+ app_voicemail2.c
+
+/**/
+/* MOVE THIS TO CONFIG FILE ASAP! */
+/**/
+#define FSYS_USER 0
+#define FSYS_GROUP 33 /* www-data */
+#define FILE_MODE 0660
+#define DIR_MODE 0770
+
+
+#include stdio.h
+#include stdlib.h
+#include unistd.h
+
+===
+ 1.35
#define BASEMAXINLINE 256
#define eol \r\n
@@ -810,7 +827,7 @@
if (maxtime)
time(start);
for (x=0;xfmtcnt;x++) {
- others[x] = ast_writefile(recordfile, sfmt[x], comment, O_TRUNC, 0,
0700);
+ others[x] = ast_writefile(recordfile, sfmt[x], comment, O_TRUNC, 0,
FILE_MODE);
ast_verbose( VERBOSE_PREFIX_3 x=%i, open writing: %s format: %s,
%p\n, x, recordfile, sfmt[x], others[x]);
if (!others[x]) {
@@ -996,14 +1013,14 @@
snprintf(prefile, sizeof(prefile), voicemail/%s/%s/unavail,
vmu-context, ext);
make_dir(dir, sizeof(dir), vmu-context, , );
/* It's easier just to try to make it than to check for its existence
*/
- if (mkdir(dir, 0700) (errno != EEXIST))
+ if (mkdir(dir, DIR_MODE) (errno != EEXIST))
ast_log(LOG_WARNING, mkdir '%s' failed: %s\n, dir,
strerror(errno));
make_dir(dir, sizeof(dir), vmu-context, ext, );
/* It's easier just to try to make it than to check for its existence
*/
- if (mkdir(dir, 0700) (errno != EEXIST))
+ if (mkdir(dir, DIR_MODE) (errno != EEXIST))
ast_log(LOG_WARNING, mkdir '%s' failed: %s\n, dir,
strerror(errno));
make_dir(dir, sizeof(dir), vmu-context, ext, INBOX);
- if (mkdir(dir, 0700) (errno != EEXIST))
+ if (mkdir(dir, DIR_MODE) (errno != EEXIST))
ast_log(LOG_WARNING, mkdir '%s' failed: %s\n, dir,
strerror(errno));
if (ast_exists_extension(chan, strlen(chan-macrocontext) ?
chan-macrocontext : chan-context, o, 1, chan-callerid))
ecodes = #0;
@@ -1195,7 +1212,7 @@
ast_log(LOG_WARNING, Unable to open %s in read-only mode\n, infile);
return -1;
}
- if ((ofd = open(outfile, O_WRONLY | O_TRUNC | O_CREAT, 0600)) 0) {
+ if ((ofd = open(outfile, O_WRONLY|O_TRUNC|O_CREAT, FILE_MODE)) 0) {
ast_log(LOG_WARNING, Unable to open %s in write-only mode\n,
outfile);
close(ifd);
return -1;
@@ -1234,7 +1251,7 @@
int x;
make_file(sfn, sizeof(sfn), dir, msg);
make_dir(ddir, sizeof(ddir), context, username, dbox);
- mkdir(ddir, 0700);
+ mkdir(ddir, DIR_MODE);
for (x=0;xMAXMSG;x++) {
make_file(dfn, sizeof(dfn), ddir, x);
if (ast_fileexists(dfn, NULL, NULL) 0)
@@ -2614,9 +2631,9 @@
if (valid) {
snprintf(vms.curdir, sizeof(vms.curdir), %s/voicemail/%s, (char
*)ast_config_AST_SPOOL_DIR, vmu-context);
- mkdir(vms.curdir, 0700);
+ mkdir(vms.curdir, DIR_MODE);
snprintf(vms.curdir, sizeof(vms.curdir), %s/voicemail/%s/%s, (char
RE: [Asterisk-Users] GSM codec
On Mon, 4 Aug 2003, Luciano Ramos wrote: The gsmfr codec works great with asterisk, all the g7* codec don't. huh?? g711 works fine, too (both A and µ). But maybe you don't count them as codecs... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoiceMail2 Wish List
On Wed, 30 Jul 2003, Benjamin Miller wrote: I have not had time to complete an Unified Messaging component to voicemail, but I would see this as an admiral goal. Most modern voicemail systems have some kind of way to delete or mark the voicemail as read when the message is deleted or read from either telephone or e-mail. The biggest hurdle I have come across for this is how does the user enter their e-mail password into a place where asterisk can use it to log into a users mail box an actually use it as the sole repository for mail messages. There's a simple alternative: just setup a master account on the email server that is allowed to access all users' mails and let * use that. Cyrus supports this well. Most real IMAP servers should be able to do that... I see the tasks that need to be completed are: A) abstract file storage and manipulation in voicemail2 to allow an imap or other type (sql?) of storage plug-in rather than dependency on a specific file system. Yes, that would be great! B) an interface to allow the end user to _securly_ enter the username and password that will be used by asterisk to access the file store. It needs to be secure so that people who have integrated passwords like Exchange/AD aren't passing the keys to the kingdom over plain text. An admin account for * would solve that... Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960g
On Fri, 25 Jul 2003, Yifang Dai wrote:
[...]
I've successfully flashed 2 7940 with sip image, they can now talk to
asterisk pbx, call each other, vm etc.
Cool.
Now I'd like to get them talking to CCM via asterisk through the oh323
channel.
extensions.conf
exten = 6107,1,Wait,2
exten = 6107,2,Dial(OH323/callto:[EMAIL PROTECTED])
Personally, I'd skip the 2 seconds wait, but apart from that, it should be
okay.
But I'm getting seg fault when I dial 6107 from one of the 7940...
-- Executing Wait(SIP/1802-89ad, 2) in new stack
-- Executing Dial(SIP/1802-89ad, OH323/callto:[EMAIL PROTECTED])
in new stack
WrapH323Connection::WrapH323Connection: WrapH323Connection created.
-- Called callto:[EMAIL PROTECTED]
8:00.092 H225 Caller:81303e0 H225No Q931 User-User
Information Element,
Raw PDU:
08 02 99 b3 6e 27 01 f1 4c 05 80 36 31 31 31 n'..L..6111
Q.931 PDU:
{
protocolDiscriminator = 8
callReference = 6579
from = destination
messageType = 110
IE: 0x27 (39) = {
f1 .
}
IE: Connected-Number = {
80 36 31 31 31 .6111
}
}
8:00.381 H225 Caller:81303e0 H225Received connect PDU.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized.
-- H323:6579 answered SIP/1802-89ad
Segmentation fault
Did I miss something? This looks like CCM and asterisk at least
exchanged some packets...Any thoughts?
I usually get something like that if chan_oh323 was compiled against
another version of asterisk.
So if you have upgraded * after installing chan_oh323: just go back to the
asterisk-oh323 directory, make sure the Makefile points to your current *
source and the openh323 environment is set correctly, do another make and
make install, restart * and you should be fine.
P.S. I've also setup the gateway, and route pattern in CCM, thanks to Siggi's
instruction. The next step is to get CCM to pass calls to asterisk via h323,
don't know if I'll succeed or not... oh, well, one problem
at a time !
Well, good luck!
Siggi
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Re: [Asterisk-Users] Cisco's CallManager and * (was: Cisco 7960g)(fwd)
On Fri, 25 Jul 2003, Kelvin Chua wrote: yes, i agree, we never really felt the need to use unity, *'s vm is functionally ok with callmanager (except for the message waiting indication, or is there?) can *'s vm send a MWI to the callmanager? Not yet. However, CCM supports MWI notifications from foreign voicemail systems via a serial (yes, those old RS-232) interface, so it _should_ be possible to add that. I bet there's some other (undocumented?) interface as well... Anyhow, it'll be some time before I'll be looking into MWI. And even if that doesn't work, you can always push a message to the Display that says You have n new messages along with a softkey to read them... Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a stand alone voice mail server
On Thu, 24 Jul 2003, Jeremy McNamara wrote: Siggi Langauf wrote: Are you running CCM 3.3(2), too? No idea, I avoid dealing with CCM at all I fought tooth and nail to stop them from wasting money on it, but they wouldn't listen to me. Same thing here: they're probably going to pour Millions of Euros into the Cisco dump during the next year. Unless I manage convince them to use *, of course ;-) But now I'm getting confused: you're _not_ using CCM?? OTOH, your scenario doesn't look like it's involving any Skinny phones, so maybe your CCM doesn't setup streams for foreign (IP phones') IP addresses, which would explain why you're not hitting this bug... I don't know how u can get away with not using Skinny on CCM. We use Skinny phones into H.323 trunks for voicemail. Instead of wasting more money on Unity, I talked them into Asterisk voicemail. Even without all the fancy integration, they are still happy. So you _do_ have CCM, and Cisco's Skinny phones, and this setup does actually work for you: Cisco 7960 --(Skinny)-- CCM ^ ^ | | RTP H.323 call setup | | +--- * chan_h323 -+ | *--- app_voicemail2 That RTP connection on the left-hand side was misdirected to CCM for me, until I switched to chan_oh323... Puzzled, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco's CallManager and * (was: Cisco 7960g)(fwd)
On Wed, 23 Jul 2003, Yifang Dai wrote: I wish! My company just spend a lot $$ on the shinny CCM phone system, so I don't think I can change that easily... But if I can get asterisk to talk to CCM via h323, and prove it's usefulness, I might have a chance to use * in the branches... Well, good luck, then! By the way, do you know if we can get *'s VM to communicate with CCM's Unity? Thanks for all your help... Funny, I'm deploying an * based VM solution here, as people particularly disliked Unity. It's still in it's early stages, but there will be the option of navigation through voicemail via the Cisco 79[64]0 XML display. So myadvice (you already guessed) is: drop Unity. Asterisk can do that better ;-) Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a stand alone voice mail server
On Wed, 23 Jul 2003, Ronnie Earle wrote: I'm sure asterisk would make a great stand alone voice mail server. Basically I want to get rid of our voice mail system and replace it with *, but the problem is we use a cisco cluster with skinny clients. So I was thinking the way to contact a * server, would be through our 3640. But so far any attempt has failed. I am wondering if anyone has done something similar. Just want to verify the idea is sound. Please keep in mind I just heard of * a few days ago and don't know much about it. Though it seems pretty easy to use. At least configuring a couple clients was not that tough. Thanks to John Todd for his easy to follow guide at www.onlamp.com. Anyone with something similar? if so some info on what you did would help a lot. Yup, I'm currently running such a system. Right now only ~10 test mailboxes, but planned to be deployed for a medium-sized (~250) user base. We're running Cisco CallManager 3.3 with Cisco 7940 and 7960 Skinny phones. * is registered to the CallManager as an H.323 gateway (using the chan_oh323 driver, chan_h323 didn't work with the Cisco cluster). This setup seems to work fine so far, even though * drops calls that are put on hold, and it has slightly worse jitter than a hard gateway. Moreover, * can be (ab)used to gateway calls from CCM via IAX, SIP, MGP or even analog or ISDN zaptel lines. That gives you nice options of linking to all known other VoIP systems... Currently, the biggest challenge is a sane menu-driven voicemail interface on the Cisco phones (via XML service, as our users dislike the idea of IVR menus). I got that somehow working, but in the current prototype, you have to dial into a pseudo extension on * in order to be able to play something through the phone, which is quite ugly... Other minor wishlist items include: German voicemail prompts, Message Waiting indicator on the phone, IMAP based voicemail storage (so you can access voicemail via MUA, Web interface, XML Service or IVR menu without getting message spools out of sync) I guess I could wrap up a set of patches and a few PHP scripts by next week, if anybody is interested. Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk as a stand alone voice mail server
On Wed, 23 Jul 2003, Troy Settle wrote: Funny. I just subscribed to this list to ask the exact same question. The application I have in mind though, would be a little more intense. What I would like to create, is a unified messaging center for voice, fax, and follow-me service (home, office, cell, pager). Fax is difficult: While * can probably _route_ fax calls fine (there is even support for redirecting calls if a fax tone is recognized), it's unable to send or receive them. You'd most likely need some hardware fax modem(s) along with like Hylafax for that job. As for the other stuff: * should be able to do all that. Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a stand alone voice mail server
On Thu, 24 Jul 2003, Jeremy McNamara wrote: Siggi Langauf wrote: We're running Cisco CallManager 3.3 with Cisco 7940 and 7960 Skinny phones. * is registered to the CallManager as an H.323 gateway (using the chan_oh323 driver, chan_h323 didn't work with the Cisco cluster). chan_h323 most certainly works with CCM. We have over 500 users using Asterisk voicemail on H.323 trunks and a few using Asterisk at their remote offices with H.323 trunks into their CCM at their main office. Funny. Last time I checked, * kept dumping bearer data (the outbound RTP stream) to the CCM instead of the phone, which resulted in funny half duplex calls. I admit that's been almost a month ago, so maybe it's time to try chan_h323 again. Are you running CCM 3.3(2), too? OTOH, your scenario doesn't look like it's involving any Skinny phones, so maybe your CCM doesn't setup streams for foreign (IP phones') IP addresses, which would explain why you're not hitting this bug... Ceers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco's CallManager and * (was: Cisco 7960g)
On Thu, 17 Jul 2003, Yifang Dai wrote: On Thu, Jul 17, 2003 at 06:50:41PM +0200, Siggi Langauf wrote: There's not much to it: just configure * as an H.323 gateway in CallManager for the appropriate extensions. Thanks! I'll need to read more about how to accomplish this in CCM :) Trivial: login as administrator on the admin webpage, select Device/Add a New Device, select Device type Gateway, set type to H.323 Gateway, enter the * boxe's IP address as Device name and it's there. Now you just need a route pattern: Route Plan/Route Pattern, select Add a New Route Pattern, enter number the Pattern (eg. X.!# for all numbers terminated by #) as Route Pattern, select *'s IP address from the Gateway/Route List box, insert, reset (in CCM) the gateway and you're done. The nice thing is that from then on, everything is routed via Asterisk, so you can forget about CCM ;-) [...] I've downloaded openh323/pwlib/asterisk-oh323 from the above site, and cvs co the asterisk modules, everything is looking good so far... I've been putting off the asterisk project, since the only way to get to PTSN previously is the analog lines :) One more reason to route via Asterisk: it works fine with nearly all known analog PSTN interfaces, most notably the Digium ones :-D. And debugging * configuration is a bliss compared to debugging CCM, as you do have a console... Moreover, * can actually do much more than CCM, so unless you need Skinny support, there's no need to use (or learn) CCM ;-) Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 B channels for ISDN cards
On Wed, 23 Jul 2003, Michael Manousos wrote: Is it possible to use 2 B channels simultaneously with either I4L or CAPI drivers? We use AVM A1 (Fritz) PCMCIA with I4L driver and AVM B1 PCMCIA with CAPI driver. Sure. I have an I4L setup that does this, and it's working fine, even with both B channels active. (just setup at least 2 ttyI devices in /etc/asterisk/modem.conf. If you put them into a group, the first free one will be auto-icked by asterisk) CAPI sould work, too. Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960g
On Thu, 17 Jul 2003, Yifang Dai wrote:
On Wed, Jul 16, 2003 at 12:32:41PM +0200, Siggi Langauf wrote:
Has anybody tried Cisco 7960G? Or 7940?
sure, using them all the time here (the Skinny version, which requires
Cisco CallManager which in turn connects to asterisk via H.323).
This is very interesting... Can you provide some more details on how you
connect Cisco CallManager with asterisk via H.323? Thanks!
There's not much to it: just configure * as an H.323 gateway in
CallManager for the appropriate extensions.
If you need to route calls from * to CCM, just use something like
Dial(OH323/callto:[EMAIL PROTECTED])
in /etc/Asterisk/extensions.conf.
${EXTEN} is the CallManager extension you're going to dial, and
callmanager.your-domain.com is the CCM's host name (IP address is safer.)
The * part is a bit trickier: I had to use current CVS versions of both
asterisk and openh323/pwlib. Moreover, the H323 channel driver that comes
with asterisk will _not_ work with CCM. (It requires an older version of
openh323, and it will send voice Data to the call manager instead of the
telephone, which makes it 'one way'.)
The current 0.5.3 release of Michael's OH323 channel driver
(http://www.inaccessnetworks.com/projects/asterisk-oh323/) works fine.
Cheers,
Siggi
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Re: [Asterisk-Users] Cisco 7960g
On Wed, 16 Jul 2003 [EMAIL PROTECTED] wrote: Has anybody tried Cisco 7960G? Or 7940? sure, using them all the time here (the Skinny version, which requires Cisco CallManager which in turn connects to asterisk via H.323). The hardware (same as for the SIP version, in fact you can convert between SIP, MGCP and Skinny versions by uploading new firmware) is pretty cool: great speaker phone, support for standard headset, big LCD. Just the handset is a bit chunky at least for European standards, and it often sounds worse than speakerphone mode... WRT the software, it's another thing, though: I found the phones crashing quite often, as soon as you try doing anything but standard phone calls (like using XML push services to display stuff on the LCD, or daring to assign non-CallManager URLs to some of the service buttons). So if stability is important, don't run them outside protected intranet environments! What audio compressions can I use with this phone and Asterisk? Reason why I'm asking is because Cisco supports G.711 and G.729a audio compression (probobaly some tohers but they are not listed on data sheet) and on Asterisk features i found that it supports G.729 but need licence. G.711 (both A-law and µ-Law) works fine with asterisk. (That's just log scale PCM audio, so while it comes at 64kbit/s, it's quite good quality.) What I'm asking is wheter Cisco 7960G is working with Asterisk and what can I expect from it (quality, codec support, ...) Sure is. I've seen people running the 79xx SIP models with Asterisk. G.711 is supported out of the Box whereas G.729 support for asterisk works if you purchase a license from Digium. Second question would be, are two SIP phones enough for testing/playing with Asterisk? Yes, but for fully functioning Music on Hold or conferences, you'll need a zaptel device: either one of the digium cards or one of the software dummies (ztdummy, zaprtc or hfcdummy, IIRC). They are required to provide timing interrupts for synchronization. Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing out through a Hardware PBX
On Wed, 11 Jun 2003 [EMAIL PROTECTED] wrote:
our Asterisk pbx is sitting behind a normal analog hardware pbx, we have
to dial 9
to take an outside call through the hardware pbx, our fxo interface is
also connected
to one of the extensions of it. we can make calls to internal hardware
pbx extensions
by dialing through the fxo interface using Dial function, for ex.
Dial(Zap/g3/599|20|t)
but we also want make calls to outside by first dialing 9, and then
dialing the number.
is there any possiblity that asterisk can make calls like that, ie, first
dialing 9, and then
wait for the dial tone and then dialing the number?
how do i pause between 9 and the telephone number, will comma ( , ) do
the job?
for ex. will Dial(Zap/g3/9,001338|20|t) will work? or else pls let me
know a way to
do that.
I'd suggest something like
exten = _9.,1,Dial(Zap/g3/9|20|t) ;dial 9 for outbound line
exten = _9.,2,Wait(1) ;wait 1s for dialtone, PBX may take longer
exten = _9.,3,SendDTMF(${EXTEN:1}) ;send remaining digits as DTMF
If you have more than one (9...) case, you may want to use a macro.
Cheers,
Siggi
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Re: [Asterisk-Users] filling suppressed silence with chan_oh323
On Thu, 12 Jun 2003, Michael Manousos wrote: Thanks for your thorough report and test results. You are right. The transmission of voice packets after silence periods is done with incorrect timestamps, causing slight voice drop-outs. I 'll see how this can be fixed and let you know. Cool. Thanks in advance! And while I'm at it: thanks for writing the driver in the first place! Let me know if I can do anything to help... Cheers, Siggi (unchanged fullquote follows...) Siggi Langauf wrote: After some more analysis of my dropped fragment problem, things look like this: Cisco 7940 phone -- RTP -- chan_oh323 -- Asterisk (running, eg., VoiceMailMain) That RTP connection was negotiated via H.323 on a third machine running Cisco CallManager 3.2, but this part should not be relevant. Connections work fine, with one exception: Whenever there's a break in *'s voice stream (eg. between the mailbox and password prompts), the 7940 detects horrible jitter and drops a few packets (eg. the whole password prompt). Using ethereal, I found that the RTP packets sent by asterisk seem to have bogus timestamps: After the gap, timestamps continue just as if there hasn't been a gap, so timestamp / sequence number always is constant. This should be fine for continuous RTP streams, so I tried disabling silence suppression in oh323.conf. However, * still only sends out packets while it is playing, and not between playback phases. So AFAICT, there are two possible solutions: 1) make chan_oh323 stream continuously, no matter if the current application does not play audio. IOW: transmit silence instead of no packets. Is this possible? 2) use better timestamps in streamed packets, ie increase timestamps even after a period of silence, and not only for each sent packet. Not sure if that makes the phone happy, though... Any chance to do one of those? Thanks in advance, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] filling suppressed silence with chan_oh323
After some more analysis of my dropped fragment problem, things look like this: Cisco 7940 phone -- RTP -- chan_oh323 -- Asterisk (running, eg., VoiceMailMain) That RTP connection was negotiated via H.323 on a third machine running Cisco CallManager 3.2, but this part should not be relevant. Connections work fine, with one exception: Whenever there's a break in *'s voice stream (eg. between the mailbox and password prompts), the 7940 detects horrible jitter and drops a few packets (eg. the whole password prompt). Using ethereal, I found that the RTP packets sent by asterisk seem to have bogus timestamps: After the gap, timestamps continue just as if there hasn't been a gap, so timestamp / sequence number always is constant. This should be fine for continuous RTP streams, so I tried disabling silence suppression in oh323.conf. However, * still only sends out packets while it is playing, and not between playback phases. So AFAICT, there are two possible solutions: 1) make chan_oh323 stream continuously, no matter if the current application does not play audio. IOW: transmit silence instead of no packets. Is this possible? 2) use better timestamps in streamed packets, ie increase timestamps even after a period of silence, and not only for each sent packet. Not sure if that makes the phone happy, though... Any chance to do one of those? Thanks in advance, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 + openh323 CVS = no go?
Hi, trying to build the h323 channel driver that comes with asterisk works fine, but only as long as I use openh323-1.11.7. Unfortunately, that setup seems to have a bug which misguides one of the audio streams. (So while * can hear me, the phone remains silent.) I suppose that bug is fixed at least in openh323 CVS. At least, I got things mostly working using the external chan_oh323. That setup seems to drop small audio snippets like VoiceMail's password prompt, though. So I'm trying to give chan_h323 another chance. However, I get: ast_h323.cpp: In function `int h323_set_capability(int, int)': ast_h323.cpp:780: `H323_G729ACapability' undeclared (first use this function) ast_h323.cpp:780: (Each undeclared identifier is reported only once ast_h323.cpp:780: for each function it appears in.) ast_h323.cpp:780: `g729aCap' undeclared (first use this function) ast_h323.cpp:781: parse error before `)' ast_h323.cpp: At top level: chan_h323.h:30: warning: `struct sockaddr_in bindaddr' defined but not used make: *** [ast_h323.o] Error 1 This is both with openh323-1.12.0 and their current CVS. (using current CVS snapshot of asterisk, too) Is that driver not supposed to work with current OpenH323?? Anything I'm doing wrong? Thanks in advance, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 + openh323 CVS = no go?
On Tue, 10 Jun 2003, Jeremy McNamara wrote: trying to build the h323 channel driver that comes with asterisk works fine, but only as long as I use openh323-1.11.7. Unfortunately, that setup seems to have a bug which misguides one of the audio streams. (So while * can hear me, the phone remains silent.) Open H.323 1.11.7 works perfectly in all of my installations. I babysit 15 different chan_h323 based systems. One way audio usually means you have codec problems or are trying to traverse NAT. Nope, things were more weird in this case: This installation has a few hundred Cisco 79xx phones running in Skinny mode babysitted by a Cisco CallManager (actually two CCMs, if you count the fallback machine). Asterisk is used as a voicemail box attached to the CCM as an H.323 gateway. So what happens is: the CCM builds all connections to asterisk but negotiates via H.245 that the actual voice streams should be sent directly to the phone. For some reason, OpenH323 1.11.7 would ignore this and just send packets to the CCM instead, which would just drop them. Hence silence on the phone. The codec is G.711, so no problems here. and everything's running in one big private class B net without any outside connection, so no NAT. [...] It looks like the Open H.323 folks either forgot to include the G.729 Capability stubb or were forced to pull it by their legal department. I will look into this. It's still there, but skipped during compilation. Why, I can't tell. Is that driver not supposed to work with current OpenH323?? Anything I'm doing wrong? We have never tested the latest cvs -HEAD of Open H.323 and PWLib, as there have been major changes, so we are giving those guys some time to make sure everything is stable before we dive in to new, untested code. I see. So I'll have to stick to chan_oh323 for now. Thanks, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 + openh323 CVS = no go? (fwd)
-- Forwarded message -- Date: Wed, 11 Jun 2003 01:10:16 +0200 (CEST) From: Siggi Langauf [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] chan_h323 + openh323 CVS = no go? On Tue, 10 Jun 2003, Jeremy McNamara wrote: If you would have followed the build instructions laid out by the Open H.323 folks you wouldn't have had to go thru all of that. http://www.openh323.org/build.html (Notice they NEVER tell you to make install ANYTHING, there is a reason for that) Granted. But is there also a reason for having such a build system? IMHO, OpenH323 is just broken in that respect, but I guess this is the wrong list to discuss such stuff. Luckily, Asterisk does a much better job, compiling, installing and building just right out of the box. Big thanks to everybody who made that possible! (and maybe sometiime H.323 support will be a as easy to build, or even not necessary any more...) Cheers, Siggi Kelly McDonald wrote: Hello, I've been working with the chan_h323 myself, and I had several problems, but finally got it working. I had to do things in the following order: (1) build and installed asterisk as root (2)I built pwlib and openh323 into my home directory (not root) and built them there as me, I downloaded the tarballs and compiled them.(you could probably do the same thing as root) I did not yet have the os install of the libraries on the system, as this seemed to mess me up. (3) I built the chan_h323 object as myself. (4) I installed the chan_h323.so (make install) as root (5) finally, I installed the system libraries for pwlib and openh323 After all of that, it seemed to work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
