Re: [Asterisk-Users] Cisco 7960 SIP Images
Wanta take a guess what would happen if Cisco decide to really enforce the legal rules? I'll bite: Their market share would plummet in all their markets, and then smaller, more innovative companies would become more able to compete with them, and the overall marketplace would be vastly improved because of more participants and more choices? B. I can't wait for that day. I don't deny that cisco make some nice products, but I don't like companies who have the attitude that since they're big and powerful they can invent whatever pricing policy they want and rip off the consumer. Of course, the argument is that as a consumer I can simply choose not to buy if I don't want to - and indeed we are now turning towards Polycom phones rather than Cisco. Cisco phones are already expensive enough - it is simply cheeky that they should have to charge further for the software that runs on the phone. That is a joke. All hardware includes software to some degree, yet one doesn't have to pay creative labs for the drivers that power their soundcards, nor Vegastream for the bundled web manage interface. And when bugs are fixed, it should be the responsibility of manufacturers to update them - the bugs shouldn't exist in the first place. Reading through some of the arguments on this thread (both pro anti Cisco) it is interesting how some feel that we should be paying Cisco the money they are demanding because it funds research and development - ironic considering this very list is about community support for a community made project. Asterisk, like many other open source projects, prove that innovation CAN and DOES take place without direct financial incentive - indeed the likes of sendmail, bind, apache etc... were around years before Microshaft came out with its equivalent tripe - and they charge piss loads for what is effectively a piece of shite. For the Cisco phones we DO have, we don't have any purchased licenses and I don't ever intend on getting any either. Cisco can sue my ass if they really want to. - Terence ___ 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] H323 in Asterisk
I have posted before but didn't get any replies so i'll ask again in a more simple way : Does H323 work on asterisk out of the box? I notice there is already a channels/chan_h323.c file, but creating an h323.conf file I can't seem to get H323 working. Do I have to compile an additional package first or something? I tried the asterisk-oh323 thing, but can't get it to compile. Terence ___ 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 7960 SIP Images
I think John's said it all - I have absolutely nothing to add! I'm just posting to second his opinion. Terence On 29 Mar 04, at 3:22 AM, John Baker wrote: -- snip -- I finally got ahold of someone at Cisco to sell me the support contract, but it took three weeks and a couple of follow up phone calls for them to process the paperwork and assign me a number. You'd think Cisco would have an easy sign up over the web for this stuff, but no. You've got to send them a check (Why wouldn't you take a credit card???) and answer a barrage of questions before you get the thing. I wondered why a company like Cisco would make you jump through so many hoops. I soon got my answer: one of their sales reps called within days to discuss purchasing more product. I'd be glad to talk to you about it, I told him, but we're a bit premature. I need to evaluate your phone with a current image and I'm getting nowhere with your technical support. Any chance you could speed up the process? It might help you get more business... No chance. After three weeks worth of runaround, I finally got my SIP image. Again the phone was nice, but the service wasn't. The price definitely wasn't. Oh, and let's not forget about the software license requirement and the power cube (purchased separately of course) Add all that up and you're paying alot for what you're getting. I went with the Polycom phones and never looked back. They're every bit as nice as the Cisco phones for a lot less money. John ___ 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] Problems with oh323-0.5.10
I am trying to get asterisk working with H323 - but I don't seem to be getting very far. I have downloaded the asterisk-oh323-0.5.10 package, but cannot seem to compile it ; I get the following errors: ... [ a lot omitted ] . wrapper_misc.cxx:72: error: parse error before `else' wrapper_misc.cxx:80: error: invalid use of undefined type `class WrapMutex' wrapper_misc.hxx:61: error: forward declaration of `class WrapMutex' wrapper_misc.cxx: In member function `void WrapMutex::Signal(const char*, int, const char*)': wrapper_misc.cxx:81: error: parse error before `::' token wrapper_misc.cxx:82: error: `Class' undeclared (first use this function) /usr/share/pwlib/include/ptlib/indchan.h: At global scope: /usr/share/pwlib/include/ptlib/indchan.h:332: error: storage size of ` channelPointerMutex' isn't known /usr/share/pwlib/include/ptlib/unix/ptlib/thread.h:167: warning: `void PX_ThreadEnd(void*)' declared `static' but never defined /usr/share/pwlib/include/ptlib/unix/ptlib/pprocess.h:144: warning: `void PXShowSystemWarning(int)' declared `static' but never defined /usr/share/pwlib/include/ptlib/unix/ptlib/pprocess.h:145: warning: `void PXShowSystemWarning(...)' declared `static' but never defined make[1]: *** [wrapper_misc.o] Error 1 I have declared my PWLIBDIR as /usr/share/pwlib , which should be correct, and I have also declared the OPENH323DIR to /usr/include/openh323 (i've tried /usr/share/openh323 too, also doesn't work). For pwlib i've currently installed ver. 1.6.3 and openh323 is 1.13.2 . These were both done through the portage system in gentoo linux. I attempted to manually compile and install pwlib 1.5.2 as suggested, but compilation of pwlib failed! Is the asterisk-oh323 package even needed? What does it do exactly? On the asterisk website I get no indication that it is required and the site talks as if asterisk should ALREADY support H323 out of the box. Indeed, in the asterisk source /channels directory there is a chan_h323.c file - does this mean it already supports H323? How does one go about getting H323 to work in asterisk? Thanks, Terence ___ 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] Unable to create vpb channel
Hi Steven, It looks to me as if you haven't defined 'channel = 9' in your vpb.conf file ... yet the log output shows that you are attempting to use that undefined channel. Try adding it to your vpb.conf first. Terence On 10 Feb 04, at 2:24 AM, Steven Kawuma wrote: Hi all, I'm using a voicetronix openswitch6 card with asterisk. When I try to dial the vpb phone from my application, I get t he following error: -- Executing Dial(Zap/1-1, vpb/1-9|10|mtT||/usr/local/sbin/parlix_dial_event 2 196) in new stack -- 1-9 requested, got: [None] NOTICE[524311]: File app_dial.c, Line 506 (dial_exec): Unable to create channel of type 'vpb' What does it mean? Below is my vpb.conf: -- snip -- ___ 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] Pictures of new multiport FXO/FXS from digum
Does the voicetronix card work with Asterisk? Yes, and no. It works in that you are able to use it to make and receive calls - but to say it works well would be an overstatement. We are currently using the OpenLine4 card and are having problems dialing (card dials too early ; doesn't support 'w' to delay dialing ; DTMF isn't recognised correctly from certain phones - namely Cisco 7960) , and also have quality problems when using more than one line simultaneously. For single port usage though, it's fine. If you don't yet have a card then I would suggest for the meantime looking elsewhere. There may be nothing wrong with the Voicetronix hardware as such, but clearly it's still got some compatibility issues with Asterisk. Terence ___ 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] How to delay dialing
Oops... sorry about that, stupid webmail system defaults to HTML (don't use it very often so frequently overlook this. I usually send mail using 'apple mail', which seems to screw up even plain text e-mails, but... well... have to retaliate against Outlook Express users some way!! Anyways - does anyone know if Voicetronix even supports the use of a 'comma' or '' even after they are successfully converted from 'w' and 'f' respectively? Thanks again. Terence I'll look at it again, but since I don't have a VoiceTronix card installed in any of my machines yet, I can't test it directly. -Tilghman ___ 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 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] Are there any list moderators?
I'm just curious... I have several times posted a message accidentally using the wrong account - since the address I use for this list isn't my default one. I often re-post using the correct account, and get a notification on the first that my message is 'pending approval'. I don't expect my wrong messages to get approved, but they don't seem to get rejected either - nor have I been sworn at by any moderators complaining about me wasting their time! ... does anyone actually moderate these such messages? Terence ___ 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] How to delay dialing
Hi Tilghman, Thanks for your reply - really appreciated! I have tried applying your patch, but unfortunately get the following compilation errors: chan_vpb.c: In function `int vpb_call(ast_channel*, char*, int)': chan_vpb.c:683: syntax error before `char' chan_vpb.c:696: `t' undeclared (first use this function) chan_vpb.c:696: (Each undeclared identifier is reported only once for each function it appears in.) make[1]: *** [chan_vpb.o] Error 1 make[1]: Leaving directory `/home/downloads/Telephony/asterisk-0.7.1/channels' make: *** [subdirs] Error 1 Looking at the patch file, am I correct in assuming then that the Voicetronix OpenLine4 does in fact support the comma and ampersand characters - but that it's just a matter of getting the asterisk's chan_vpb to translate the use of 'w' and 'f' to ',' and '' respectively for the Voicetronix API? Thanks again, Terence 1) The 'W' character is only for the zaptel channel. 2) It's case insensitive (i.e. it does NOT need to be uppercase). See line 2387 of zaptel.c if you'd like to confirm this for yourself. 3) There is no current way within Asterisk to insert a pause into the Voicetronix driver. 4) There is no current way within Asterisk to insert a flash-hook into the Voicetronix driver. 5) The solution for 3 and 4 is attached. This patch will allow you to use the 'w' OR the 'W' character to insert a pause and to use the 'f' or 'F' character to insert a flash-hook. Please note (VERY IMPORTANT): in the Voicetronix driver, the pause is 1.0 seconds, not 0.5 seconds, like it is in the Zaptel driver. -Tilghman ___ 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] Junk calls from FWD numbers
Nope... we're not getting junk calls on our number either! Terence On Wed, 28 Jan 2004 22:01:44 -0700 (MST), Greg Hill wrote On Tue, 27 Jan 2004, Chris Albertson wrote: Question: Does everyone with an FWD number get these junk calls or am I the only lucky one? I just got an FWD number a couple days ago, but haven't had that experience yet. And no, I haven't tried calling you to see if you'd answer. :) Greg ___ 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] DTMF wrongly recognised
Before I start ... I know there's been a lot of talk over the last couple of
days about the list being slow etc... I understand there were problems and
that mail is starting to get through now. Just wondering - should I be
surprised if something I sent 48 hours ago hasn't turned up at all? And
something I sent about 6 hours ago still hasn't turned up (yet). I'm just
curious that's all - hope my account isn't blacklisted or something!
Anyways, if my post from 48 hours ago gets through then please ignore it -
i've solved that problem. However, for some reason now that I can make
incoming outgoing PSTN calls without a delay it is common to end up dialing
the wrong number. Or rather, you would dial a number - and reach someone
random at a different number.
My PSTN is a Voicetronix OpenLine4. My vpb.conf file is set to use fxo mode
(I gather nothing else works for the OpenLine4). For DTMF in sip.conf , users
are set to use rfc2833 . I tried inband but that again crashes voicetronix
driver.
My dialplan is 'exten = _9.,2,Dial(vpb/1-2/${EXTEN:1},60,m)'
What happens now is that to dial on my Cisco 7960, I would typically have to
Dial 99# to get Asterisk to make the call - even though actually it's just
patching me through to a dial tone. I then manually dial the number I want to
dial relying on the DTMF from the phone to be recognised by the telco - hence
the probable error I have.
I dial like this because if I dial the full number (e.g. 923837654# for
23837654) - the Voicetronix card dials so quickly after going off hook that
the telco doesn't even recognise the first few digits!
1. How can I pause dialing so that it doesn't dial so quickly?
2. Even my archaic method above didn't cause problems before - DTMF was
recognised when there was a 2 second delay. Any way to improve DTMF
reliability (say, for auto attendants?)
Thanks!
Terence
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Re: [Asterisk-Users] 7960 Problems
We have personally found the version 6.0 Cisco images to be a big buggy... we tried them on our 7960's but this resulted in erratic behaviour as a result - although we did at least manage to upgrade it, so this isn't exactly related to your problem. If you can though, I would recommend either trying version 6.1 or sticking to 5.x - those seemed to work best for us. Terence This is not specifically related to * but * is the software I'm using so here goesS Does anyone have the correct file set for a 7960?? I've been trying to get the release 6 SIP load on one I have without any luck. The phone keeps getting the same 2 files from the tftp server and starting over. If you have the files - other than the POS30600.bin which I know is licensed - could you please send them to me so I can figure out if it's my files or my phone?? I really would appreciate any possible help with this. Thanks, Lane Hoskins, MCP Network Engineer 540.767.7626 ___ 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 7960 Problems
We seem to get these quite frequently - but as far as I can tell, it also doesn't appear to be causing any major problems as far as calls are concerned... so thus far I have just ignored them. Are the calls not actually being terminated properly by the phone, despite the user hanging up? We did have a problem with our Cisco phones such that any simple 'reload' of asterisk would cause a phone not to ring unless it re-registers - but that seems to be sorted now (for no reason ; I didn't "fix" anything!). If you aren't finding any problems with these errors either... then I would suggest just ignoring them. - unless anyone could explain what they actually mean? Terence Has anyone ever seen these errors generated by a cisco 7960? none of our other brand phones seem to generate these erros: Jan 27 21:54:07 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)Jan 27 21:54:08 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)Jan 27 21:54:12 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)Jan 27 21:54:14 WARNING[-1147556944]: chan_sip.c:2485 __transmit_response: Unable to determine sequence number from ''Jan 27 21:54:18 WARNING[-1147556944]: chan_sip.c:2485 __transmit_response: Unable to determine sequence number from ''Jan 27 21:54:22 WARNING[-1147556944]: chan_sip.c:2485 __transmit_response: Unable to determine sequence number from '' Thanks! Any feedback would be appreciated :) Chris
RE: [Asterisk-Users] G.729 Licenses from Digium
OK - but what counts as a SCSI system? These days there are lots of pseudo-SCSI systems around - such as our server which runs a serial-ATA RAID but the driver is loaded as a SCSI device. Is that still IDE? Or SCSI? Terence I know one thing for sure... G729 WILL NOT WORK after installation *(it never realy installs but does the segmentation faults), * will not start, and you will need to prevent g729 module from Starting in order for * to start. So do not buy if your box is SCSI in any part. Ta SJ ___ 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] Voicetronix OpenLine4: disable answering on a particular channel delay before dial
Hi Daniel,
Thanks for your reply - I appreciate it. I will try
the things you suggested- the suggestion to use a separate context for
inoming calls on line two seems to be OK. The physical phone line is not plugged
in at the moment (need to return to the office to do that) but all the other
tests seem to be ok.
I have one question however. You
say:
Try to insert a comma "," before the number you
dial.
Considering that my dial string is
currently:
exten =
_9.,1,Dial(vpb/1-1/${EXTEN:1})
- where would I insert the comma? I have tried
inserting it before the $ but that didn't seem to make much difference. Would
the comma be treated by asterisk as a separation of parameters? Or is it
actually interpreted as a pause?
Thanks
Terence
[Asterisk-Users] G729 - how many needed?
I have purchased a single G729 license - however, how many are actually needed? All my IP phones have G729a codecs built in (Cisco 7960 / Zultys ZIP2) - I would have assumed that if the phones can do it, and canreinvite=yes, then the phones shouldn't need to go through asterisk anyway? For calls that do go through asterisk, is a single license required for each side of the stream? (i.e. a connection between two phones needs two licenses?) I bought it hoping it could solve some of my bandwidth related problems but I seem to have no improvement in quality - so I don't know whether it is because the codec is simply not being used (one is not enough), or whether it is, but just isn't making a difference. Typically - does one need two licenses per call? Or is one enough? (The only thing I seem to successfully use the G729 license for at the moment is 411 on FWD) Terence ___ 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] Voicetronix OpenLine4: disable answering on a particular channel delay before dial
Hi there, After a lot of valuable insights from the list, incoming and outgoing calls finally work through OpenLine4! Thanks for all the input! Now I have 2 minor issues: Sometimes Voicetronix dials too quickly before an actual dial tone is obtained from the phone company. E.g. Voicetronix picks up a line and then dials immediately, whereas actually it took the phone company may be half a second to actually make the line available to gave a dialtone. As a result? 90% of the time, the first digit dialed was not received by the phone company. Is it possible to tell voicetronix to wait a second or two before dialing? Secondly, I have a phone line plugged into channel 2 that I don't want Asterisk to answer. I only want ASterisk to use it to dialout. So I need to configure Asterisk somehow to ignore incoming calls on channel 2. Is this possible? Thanks! Terence ___ 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] Codec problems (SIP)
Hi again, I've finally got Voicetronix OpenLine4 working so am a happy man ... thanks again to all those who helped! Just a few outstanding questions of curiosity : 1. I have finally got my setup to work by allowing ONLY g711alaw and nothing else. Why should enabling a few extra codecs cause problems? Surely if two phones are able to work at g711alaw, and either side had a compatibility problem with anything else (i.e. g729a at one end but not at the other) - they would automatically negotiate to use g711alaw anyway? Is the system/phones not smart enough to do this and I have to explicitly specify what everything should use? Secondly, also regarding codecs - I don't understand this as, surely, I have already enabled g729a and ulaw ... how can it complain that it can't transmit in that format, or that it can't find a path? How do you got the g729 codec? * does not include it. You must to pay for that. ... okay, fine. But where can I buy it? And is there something specific I have to buy, or does any old thing work with asterisk? Or...? Thanks again! Terence ___ 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] Playing background message
Sorry for the fragmented messages from me - one last thing I forgot to ask in my last post. When incoming calls come to us, our PSTN line is picked up almost immediately - and then asterisk will proceed to dial the SIP extensions. During this time the caller hears dead slience - obviously not very good as some would think the line just went dead and hang up. I have toyed with the idea of playing a 'welcome... your call will be answered shortly' etc... message, but can't get it to work how want it. The caller will hear a recorded message, followed by music. What I want is the caller to hear this WHILE the SIP phones are ringing - but using the 'Background' option in extensions.conf seems to make it so that my SIP phones won't be dialled until AFTER the music clip is finished - i.e. pointless. How do I truly set a background audio to play while the internal phones are ringing? Is this possible? Music on hold perhaps? Thanks, Terence ___ 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] Playing background message
Thanks for the replies I've decided to simply add 'm' to the dialplan for now, but i'll investigate call queues later - this sounds like the ideal setup for me though. For the meantime though, music on hold works fine! Thanks again. Terence I agree, I'd rather have the caller hear ringing instead of MOH as ringing gives the caller some feedback as to what is happening. I'd save the music until they've talked to someone or heard a message and are put on hold or get dumped into a queue. -Lance ___ 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] Zultys Zip2 (SIP)
Hi, Thanks for that. I did try updating the ZIP2 firmware - don't know if it helped or not, but I am able to login now by setting absolutely all settings to be the desired username - including the 'extension' number, which I just entered text for. This is very stupid though. If all these manufacturers are producing things to so-called SIP 'open standard' - why should there be so many inconsistencies in how things are done? Anyways, the important thing is it works now. Terence Hello, I don´t have any Zultys ZIP2 but I have several of Zultys ZIP4x4 and they are working great with asterisk. And I´m calling in/out without problem with chan_capi. Do you have the latest firmware in the ZIP2? They have recently changed something regarding authentication in the ZIP4x4 firmware. ---JanM--- ___ 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] Codec problems (SIP)
Hi again, Thanks for your help. Unfortunately that did not seem to solve the problem. After a bit of fiddling around, this is what i've managed to achieve with my asterisk setup so far. 1. With allow=all in sip.conf, nothing seems to work - not even voicemail. The following is sample output: Executing Ringing(SIP/TerenceParker-1af0, ) in new stack -- Executing Wait(SIP/TerenceParker-1af0, 2) in new stack -- Executing VoiceMailMain(SIP/TerenceParker-1af0, ) in new stack -- Playing 'vm-login' (language 'en') WARNING[278546]: File app_voicemail.c, Line 2707 (vm_execmain): Couldn't read username == Spawn extension (sip, 86, 3) exited non-zero on 'SIP/TerenceParker-1af0' - Why should this happen? Surely with everything enabled, any coded should work! 2. With disallow=all ; allow=alaw ; allow=ulaw ; allow=g729 ; allow=gsm (and i've also tried without some of those and various combinations): Executing Ringing(SIP/TerenceParker-af02, ) in new stack -- Executing Wait(SIP/TerenceParker-af02, 2) in new stack -- Executing VoiceMailMain(SIP/TerenceParker-af02, ) in new stack -- Playing 'vm-login' (language 'en') NOTICE[278546]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[278546]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from GSM to G729A WARNING[278546]: File chan_sip.c, Line 1182 (sip_write): Asked to transmit frame type 4, while native formats is 256 (read/write = 4/2) WARNING[278546]: File file.c, Line 521 (ast_readaudio_callback): Failed to write frame NOTICE[278546]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ULAW to G729A WARNING[278546]: File file.c, Line 170 (ast_stopstream): Unable to restore format back to 4 WARNING[278546]: File app_voicemail.c, Line 2707 (vm_execmain): Couldn't read username == Spawn extension (sip, 86, 3) exited non-zero on 'SIP/TerenceParker-af02' - I don't understand this as, surely, I have already enabled g729a and ulaw ... how can it complain that it can't transmit in that format, or that it can't find a path? 3. With the default settings (i.e. no allow OR disallow clause) normal IP to IP calls work fine. Calls to voicemail also works fine with no problems. However, PSTN calls through my Voicetronix card or calls routed through FWD fail to work. This is what happens when I dial out with my voicetronix card: Executing Dial(SIP/TerenceParker-22f3, vpb/1-1/18501) in new stack Read_channel ## vpb/1-1: Setting record mode, bridge = 0 -- 1-1 requested, got: [vpb/1-1] -- Calling 1-1/18501 on vpb/1-1 Read_channel vpb/1-1 (state=0), res=0, bridge=1 Read_channel vpb/1-1 (state=0), res=0, bridge=1 Read_channel vpb/1-1 (state=0), res=0, bridge=1 Read_channel vpb/1-1 (state=0), res=0, bridge=1 Read_channel vpb/1-1 (state=0), res=0, bridge=1 -- VPB Calling 1-1/18501 [t=0] on vpb/1-1 returned 0 -- Called 1-1/18501 WARNING[278546]: File chan_sip.c, Line 1182 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame -- vpb/1-1 is ringing WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame Read_channel vpb/1-1 (state=0), res=0, bridge=1 Read_channel ## vpb/1-1: Setting record mode, bridge = 0 WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame Read_channel vpb/1-1 (state=5), res=0, bridge=1 WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame Read_channel vpb/1-1 (state=5), res=0, bridge=1 -- Event [12=>[00] Loop Drop ] on vpb/1-1 -- vpb/1-1 handle_owned got event: [12=>0] -- handle_owned: putting frame: [-1=>0], bridge=(nil) WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame Read_channel vpb/1-1 (state=5), res=0, bridge=1 WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame Read_channel vpb/1-1 (state=5), res=0, bridge=1 -- Event [102=>[00] Dial End ] on vpb/1-1 -- vpb/1-1 handle_owned got event: [102=>0] -- handle_owned: putting frame: [4=>4], bridge=(nil) -- vpb/1-1 answered SIP/TerenceParker-22f3 -- hangup on vpb (vpb/1-1) Read_channel vpb/1-1 (state=5), res=0, bridge=1 Read_channel vpb/1-1 (state=6), res=-1, bridge=1 Read_channel vpb/1-1 terminating, stopreads=1, owner=yes -- Hungup on vpb/1-1 complete == Spawn extension (sip, 918501, 1) exited non-zero on 'SIP/TerenceParker-22f3' - again, it complains about codecs. So, at the moment, I am utterly confused! Any help would be gratefully appreciated. Terence On 13 Jan 04, at 1:39 AM, Jorge Mendoza wrote: Try in sip.conf: disallow=all allow=alaw allow=ulaw allow=gsm (in that order) I never tried with FWD Jorge
Re: [Asterisk-Users] This newbie gives up for now - sadly
Though slightly off-topic, I was wondering if anyone would have any ideas to the following regarding our Cisco 7960's. To keep this short - the plan facts: - With phone configured for NAT, works fine with Pulver FWD service from any location (home, various peoples offices etc...) BUT - ... phone does not work in my office. Cannot log on to system. This is with both Real IP AND NATed IP. - Yes I turned off NAT when testing phone with the real IP. Still didn't work - We have two incoming ISP lines in our office. Both have real IP's. No combination works with both lines. - Zultys Zip2 phones however seem to work fine with Real IP's from our office (ZIP doesn't support NAT), on both lines - MSN messenger also works fine For some reasons , our Cisco phones are just cursed when used in our office... I have no explanation for its erratic behaviour at all. Perhaps I should call in a Feng Shui expert? Terence (yes - it's a good looking phone though) As Robert's colleague that owns 7960s I can go on about the superiority of the Cisco phone. The most immediate difference is the look and feel. Everyone that has seen or held my phone says that it is nice. Everyone that picks up a Grandstream phone or looks at one says they are cheap. Grandstream should really consider putting some lead weights in the handset. Hell, the free USB phone from Voiceglo feels better than the Grandstream phone... and that is just the exterior... As soon as they get their problems with SIP functionality and stability sorted, they should spend some time and effort on product design. I understand they are trying to be competitive but people expect a phone to look and feel a certain way. cameron. ___ 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] Oops!
Didn't realise that replies are still tagged to specific threads in the mail headers. Oops! A few of my postings so far have been replies (to save me retyping the list address) - but aren't really replies (they are completely off topic). Hope this doesn't cause too many problems in the archives! But... at least now I know! Terence ___ 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] Problem registering FWD
Have a look at http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions If you sip client is behind firewall you will not be able to connect to FWD. However you can get around by using IAXTEL. check out this page: www.iaxtel.com/setup.html David Kwok Thanks for that. Actually, my machine has both an internal IP and a real IP address, so I didn't think I would need to turn on the NAT settings. But, stupidly, I set the binding interface of SIP to my internal address only - which probably explains my problems. I have now changed the binding back to 0.0.0.0 and all is working. (Well... the 'logging on' part anyway - i'm still having call problems bridging calls with an SIP phone, but that's another matter) Thanks again! Terence ___ 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] Problem registering FWD
I seem to have a problem registering my Asterisk box with the FWD service - I have the following in my sip.conf file: register=74928:[EMAIL PROTECTED]/74928 [fwd.pulver.com] type=friend secret= username=74928 host=fwd.pulver.com This this look wrong to anyone? It looks fine to me! Unfortunately, I get the following re-occuring message: NOTICE[245776]: File chan_sip.c, Line 2837 (sip_reg_timeout): Registration for '[EMAIL PROTECTED]' timed out, trying again Does anyone else have any problems with FWD? Terence
[Asterisk-Users] Zultys Zip2 (SIP)
Has anyone ever tried getting a Zultys ZIP2 phone to work with Asterisk? We have a few of these lying around in the office but are having difficulties getting them to dial out - authentication error. Curiously though, the Zip2 initially logs in correctly and is still able to receive calls. The error I get when a ZIP2 phone attempts to dial: NOTICE[245776]: File chan_sip.c, Line 4802 (handle_request): Failed to authenticate user Testsip:[EMAIL PROTECTED];user=phone>;tag=178bc-2289b WARNING[245776]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 100 (Response) WARNING[245776]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 100 (Response) NOTICE[245776]: File chan_sip.c, Line 4802 (handle_request): Failed to authenticate user Testsip:[EMAIL PROTECTED];user=phone>;tag=178bc-2289b WARNING[245776]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 100 (Response) WARNING[245776]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 100 (Response) I have entered the username/password correctly, but my guess is the phone probably does its authentication in a different way. Zultys claim the phone is fully SIP compliant and very generic - but I have been increasingly finding them working well only with their own gateways (which are extremely expensive). For those not familiar with the Zip2, I have two screenshots showing the relavent web based configuration screens - you can find these at http://parker.com.hk/zultys (don't want to fill up your mailbox with large attachments). I have tried several combinations, including using AES for the authentication too - but with no luck. Could it be that this phone just doesn't work with asterisk? Thanks! Terence
[Asterisk-Users] Voicetronix OpenLine4
Greetings...
I am trying to get Asterisk working with a Voicetronix OpenLine4 card.
Searching through the archives it seems that some have managed this
before - although I get the impression it's not a widely used card and
its use is not highly documented.
My vpb drivers were compiled correctly, and are loaded with the system
on startup. Running 'dmesg' will show:
vpb: major = 254
s[3] = V 0x56 s[2] = 4 0x34
check_region ret = 0
vpb: Manufactured 00/00/
vpb: Card version 00.00
vpb: Serial number
1 V4PCI's detected on PCI bus
I have tried the test programs bundled with the voicetronix drivers and
these managed to address the card without any problems (such as
recording the BUSY and DIAL tones).
What's strange is that when I run the programs 'vpbconf' and 'vpbscan'
I get the following output:
VPBCONF:
---
Cards detected:1
BOARD 1
vpb_pconf[0][0] = 0
vpb_pconf[0][1] = 0
vpb_pconf[0][2] = 0
vpb_pconf[0][3] = 0
vpb_pconf[0][4] = 0
vpb_pconf[0][5] = 0
vpb_pconf[0][6] = 0
vpb_pconf[0][7] = 0
vpb_pconf[0][8] = 0
vpb_pconf[0][9] = 0
vpb_pconf[0][10] = 0
vpb_pconf[0][11] = 0
MODEL : VPB4
DATE : 00/00/
REVISION : 00.00
SERIAL NUMBER :
STATIONS[1]:
TRUNKS[1]: 0 1 2 3 4 5 6 7 8 9 10 11
VPBSCAN:
---
CARD1:UNKNOWN:irq=22 sub=56345654
BOARDS:1
- is this normal? It seems to me that everything about my card is
unknown. The serial number, revision number, and date are all not
available. Surely this should not be the case? Do I have a firmware
that is too old?
When actually running asterisk, it almost seems to me as if it is just
not interfacing the hardware - when I attempt to make a dial to a PSTN
number, I get:
Executing Dial(SIP/TerenceParker-26d0, vpb/1/1/26058133) in new
stack
-- 1 requested, got: [None]
NOTICE[245776]: File app_dial.c, Line 499 (dial_exec): Unable to create
channel of type 'vpb'
== Everyone is busy at this time
-- Executing Congestion(SIP/TerenceParker-26d0, ) in new stack
== Spawn extension (sip, 926058133, 2) exited non-zero on
'SIP/TerenceParker-26d0'
1 requested, but got none? Does this mean it's not even finding my
hardware? However, note that when I change vpb.conf to something
ridiculous (such as port 5 on a 4 port card) then asterisk will
complain on startup - so evidently it notices something!
My vpb.conf file currently reads:
[interfaces]
echocancel = on
board = 1
context = sip
; Note that V6PCI channel numbers start at 7!
mode = fxo
channel = 1
In extensions.sip, I am using the following within the [sip] context:
exten = _9.,1,Dial(vpb/1/1/${EXTEN:1})
exten = _9.,2,Congestion
Also worth noting that is strange, is that if I pick up an analogue
phone while asterisk is running (an analogue phone that is connected to
the same phone line, but not to the voicetronix card directly), the
OpenLine card for some reason picks this up - and then when you input
DTMF asterisk crashes. Surely this shouldn't happen?
For our phones, we're using SIP.
If any one can think of any suggestions to address any of these
problems, please let me know - I appreciate any comments received.
Thanks!
Terence Parker
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Re: [Asterisk-Users] Cisco to Cisco - poor quality
I have managed to find time to have another go at the Cisco phones - alas, I am still having problems with Cisco to Cisco calls. Just to re-cap (it's been a few days!) i'm using Cisco 7960's and have tried setting both phones to different codecs (tried default g729a, g711alaw, and g711ulaw). Also, the other observations that have been made: - Problem is one-way. One side hears me clearly ; I don't hear the other side clearly at all (5% audible only). - Calls to MSN are fine (two way conversation is crystal clear) - Calls to a Zultys Zip2 SIP phone is also perfectly clear. - All these three tested over the same network and same VPN (call between Hong Kong and USA). - Cisco to Cisco calls worked fine with Vocal. If Cisco is able to talk fine with other devices, there should not be a problem with bandwidth or my network. However, I am finding it quite bizzarre that Cisco is unable to talk to itself. The problem shouldn't be VAD or the like - even if I talk non-stop, or the other guy does, I get the same problem. I attach a copy of my Cisco phone configuration for reference. I have even recently upgraded my phone firmware - but no luck. Platform : Cisco IP Phone 7960 Elasped Time: 08:11:26 dhcp_server : 192.168.8.254 my_ip_addr : 192.168.8.83 subnet_mask : 255.255.255.0 defaultgw : 192.168.8.254 dyn_dns_addr_1 : 0.0.0.0 dyn_dns_addr_2 : 0.0.0.0 dns_addr : 205.252.144.228 dns_backup_1: 202.14.67.4 tftp_addr : 192.168.0.252 dyn_tftp_addr : 0.0.0.0 my_mac_addr : 0007:50ac:6932 domain_name : deltapath.com my_name : SIP000750AC6932 Status Flags : 1230 image_version : P0S3-05-3-00 FirmLoadID : PC03A300 network_media_type : Auto network_port2_type : Hub/Switch tos_media : 5 phone_label : DELTAPATH tftp_cfg_dir : ./sip_phone/ phone_password : ** phone_prompt : SIP Phone language : english sntp_mode : DirectedBroadcast sntp_server : stdtime.gov.hk time_zone : HST dst_offset : 0 dst_start_month : April dst_start_day : 0 dst_start_day_of_week : Sun dst_start_week_of_month : 1 dst_start_time : 02 dst_stop_month : Oct dst_stop_day : 0 dst_stop_day_of_week : Sunday dst_stop_week_of_month : 8 dst_stop_time : 2 dst_auto_adjust : 0 time_format_24hr : 1 date_format : M/D/Y nat_enable : 0 nat_address : voip_control_port : 5060 start_media_port : 16384 end_media_port : 32766 sync : 1 xml_card_dir : xml_card_file : CARD.XML telnet_level : 2 services_url : directory_url : logo_url : http://deltapath.com/logo.bmp; http_proxy_addr : http_proxy_port : 80 enable_vad : 0 dial_template : dialplan callerid_blocking : 0 anonymous_call_block : 0 autocomplete : 1 messages_uri : 86 dnd_control : 0 preferred_codec : g729a dtmf_outofband : avt dtmf_avt_payload : 101 dtmf_db_level : 3 dtmf_inband : 1 line1_name : TerenceParker line2_name : 74xxx line3_name : 74xxx line4_name : line5_name : line6_name : line1_authname : TerenceParker line2_authname : 74xxx line3_authname : 74xxx line4_authname : UNPROVISIONED line5_authname : UNPROVISIONED line6_authname : UNPROVISIONED line1_shortname : Asterisk line2_shortname : FWD-74xxx line3_shortname : FWD-74xxx line4_shortname : UNPROVISIONED line5_shortname : UNPROVISIONED line6_shortname : UNPROVISIONED line1_displayname : TerenceParker line2_displayname : 74xxx line3_displayname : Terence Parker line4_displayname : line5_displayname : line6_displayname : proxy1_address : 192.168.0.254 proxy2_address : fwd.pulver.com proxy3_address : fwd.pulver.com proxy4_address : proxy5_address : proxy6_address : proxy1_port : 5060 proxy2_port : 5060 sip_retx : 10 sip_invite_retx : 6 timer_t1 : 500 timer_t2 : 4000 timer_invite_expires : 180 timer_register_expires : 3600 proxy_register : 1 proxy_backup : UNPROVISIONED proxy_emergency : UNPROVISIONED proxy_backup_port : 0 proxy_emergency_port : 0 outbound_proxy : outbound_proxy_port : 5082 nat_received_processing : 0 mwi_status : 0 call_waiting : 1 user_info : none cnf_join_enable : 1 remote_party_id : 0 semi_attended_transfer : 1 call_hold_ringback : 0 Thanks for any help! Terence I have never used Cisco phones, but I have had problems in the past relating to * RTP talking to a widget with VAD turned on. * RTP stack can not run on its own. It relies on receiving RTP packets for doing its timing. A simple test is to sniff the line to make sure the phones always send packets. If you see pauses, you may need to disable some type of VAD setting on the phone. Or just never quit talking when using the Cisco phone. Terence Parker wrote: I have set canreinvite=no in the sip.conf for each user (well, there are only two) using a cisco phone. What does this imply? As for whether the problem is due to the phones or asterisk however, indications would suggest both, because: - Voicemail works fine (and is clear) - I can initiate a call between MSN and Cisco, and that would sound fine. This might suggest a problem with my phones. However : - When using Vocal previously, Cisco to Cisco conversation was fine. This has
Re: [Asterisk-Users] 911 and lawsuits
It's just as well that here in Hong Kong employers don't have to worry about being sued by their staff tripping over their own laces ; or microwave oven manufacturers getting sued by old ladies drying off their poodle ; or supermarket owners getting sued by stupid customers who trip over their own kids. In most countries cases such as these would be thrown out the minute they are filed. Of course, these are slight exaggerations insofar as asterisk is concerned - because being able to dial 911 (or 999 as it is in this part of the world) is a much more 'genuine' problem. But nonetheless, it should be the responsibility of the implementor of such a system to ensure that there are adequate measures taken against system failure - such as UPS, or even a primitive analogue phone line somewhere in the home/office. Though I cannot possibly comment regarding 'fear of being prosecuted', simply because I have no reason to fear (i'm not under jurisdiction of a ridiculous judicial system) - I would say that it is a huge shame that a group of people all with the common goal of contributing towards free software projects such as this should even have to worry about things such as lawsuits. If there are people out there who have problems with asterisk, I suggest they just don't use it. To go as far as suing - that is just taking the piss! (sorry, can't think of equivalent non-British term). Terence Just curious if any of the Asterisk installers are doing anything special to protect themselves from a possible lawsuit caused by 911 failure during a Asterisk/computer crash? I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. What do you think most PBXs are? Maybe not a x86, but it is a computer. Anybody requiring customers to acknowledge and sign any kind of waiver? Just the legal fees of defending yourself in a lawsuit could sink most Asterisk installers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to Cisco - poor quality
I have set canreinvite=no in the sip.conf for each user (well, there are only two) using a cisco phone. What does this imply? As for whether the problem is due to the phones or asterisk however, indications would suggest both, because: - Voicemail works fine (and is clear) - I can initiate a call between MSN and Cisco, and that would sound fine. This might suggest a problem with my phones. However : - When using Vocal previously, Cisco to Cisco conversation was fine. This has led me to be completely stumped! I notice some mention elsewhere about asterisk lacking certain codecs because of license restrictions? Is this anything to do with me? Or should the phones still - in theory - be able to talk to each other without any problems? I have tried the cisco phone on both g729a and g711ulaw. I'm currently *trying* to get ahold of an updated firmware for my phone. I will see if this fixes the problems. Thanks again, Terence -- How are the phones talking to each other? Directly, or through asterisk? (canreinvite=what? in the sip.conf for each of them?). What I'm trying to get at here is, it is a problem between the phones, or are you having a problem possibly with the asterisk box? Some other things to know: are you running voicemail yet? If so and you can dial into it from either of the phones, how does it sound? If not, how about anything from the * boxlike the demo annoucment stuff? Daryl - Thanks for the replies. My cisco firmware is only POS3-04-2-00, though it is SIP. It used to work fine under vocal though - which was strange. Is this definitely nothing to do with asterisk? I do note however that my firmware is fairly old... except cisco aren't exactly generous with firmware upgrades. I have tried both g729a (default on my phone) and g711ulaw with no success. But i'll have another fiddle and try to get it to work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to Cisco - poor quality
Thanks for the reply. The switch is indeed a full duplex 10/100, and we have a relatively small network with low office traffic so that shouldn't be a major problem in my case. Also, our cisco phones did work under vocal (except vocal is overall rather naff) so that shouldn't point to a problem with the network infrastructure. We have eliminated all viruses too (didn't have any - and yes, I hate PC's also). I haven't got round to enabling tftp yet to enable telnet on my cisco phone, so can't get the settings just this minute. But I will soonish and then I can send it off for people to look at. Currently, everything is configured directly on the phone - I take it this shouldn't be a problem? Terence see if you can upgrade to firmware 4-3 or 4-4 another point to note, are you using a full duplex 10/100 switch? if so, you should have 'Port1 Full 100' for full duplex 100Mbit under the 'Network Statistics' If you like to email me your config settings, I will check them against our phones. telnet to the phone, and capture 'Phone show config' Doug ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco to Cisco - poor quality
I am just starting to deploy asterisk in our office to use as our primary phone system - we plan to use a Voicetronix OpenLine4 card as our PSTN gateway - but one thing at a time... haven't got that far yet. Currently, i'm trying simple IP to IP calls within the office using our Cisco 7960's phones running SIP. When I make a call between these two phones, the conversation is of a quality so bad that it is barely audible (5% makes sense). I recall having this same problem when I tested asterisk briefly one year ago. However, I did also try on this occasion to make a call between the cisco phone and MSN - that worked fine. So it would seem that the cisco phone is to blame? - but why? Does anyone know why two phones of the same type should have so much problem talking to each other? Thanks! Terence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to Cisco - poor quality
Thanks for the replies. My cisco firmware is only POS3-04-2-00, though it is SIP. It used to work fine under vocal though - which was strange. Is this definitely nothing to do with asterisk? I do note however that my firmware is fairly old... except cisco aren't exactly generous with firmware upgrades. I have tried both g729a (default on my phone) and g711ulaw with no success. But i'll have another fiddle and try to get it to work. Thanks again. Terence what firmware are you using? is it SIP? to check, push settings then status and firmware you should have a load ID like this 'POS3-04-4-00' also check the preferred CODEC we use g711ulaw as the default -- snip -- You must be doing something wrong (maybe codec problems), because I've had absolutely no problems with Cisco to Cisco calls, and I've got almost 50 deployed across the company. (For what it's worth, I'm using the ulaw codec.) Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
