Re: [asterisk-users] Cut offs on outgoing SIP calls
Hi! I've suffering cut offs after 6 or 7 seconds a call is answered, incoming calls are working fine, but outgoing ones show the gollowing messages when are being dropped [...] It seems the SIP ACK is not being received properly. I can confirm this issue: In my case it happens with calls coming in from a patton ISDN gateway to Asterisk 1.8.20.1. The calls is processed and passed to a snom phone, audio flows fine for a few seconds, but then Asterisk terminates the call. Interestingly this never happens on internal calls (from snom to snom). Downgrading to Asterisk 1.4 makes the issue go away as well. Have you tried 1.8.22? I haven't yet, but it seems to come with a fix for a deadlock in the SIP channel which *might* solve the issue we are both experiencing (see ASTERISK-21389). Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make the sum of a ${VARIABLE} + 1 ??
exten = s,n,Set(vgLabel=vg(${number}+1))
exten = s,n,GoTo(${vgLabel})
But in stead of vgLabel becoming the SUM of 2 numbers, it is just a
string :
Use the MATH function.
Philipp
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Re: [asterisk-users] Issue with asterisk
Hi! It is causing an issue for me. One SIP UA works fine - ring, forward, etc. While the other does not. Make the UAs listen on different ports (for example 5060 and 5062) and see if that solves your problem - if you can't make them have different IPs, that is. Also be sure to fully understand what the insecure= settting in sip.conf does, if you are not then do some reading on that. If you really want to use insecure then you might want to consider to combine that with permit= and deny= to improve on security. You most likely have two SIP UAs that use the same IP, of which the 6839 account is listed last in sip.conf while 3169 is trying to auth (unsuccessfully). The settings that you sent to the list mention neither 6839 nor do they mention 3169. Please state which of the SIP account those are using, on which IP (and port) these phones are. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk, VoIP and Samsung iDCS100
Hi! 2. Add BRI card(s) to the computer to run Asterisk and somehow hook up the Samsung. Do-able. Connect Asterisk to your ISDN2, then host the Samsung off the asterisk box. But then, might as well dump the Samsung and just put VoIP phones on everyones desks. If you decide to go down this route: You would need a 4-port ISDN card (since 2-port ISDN cards are hard to come by) in order to allow for direct bridging of ISDN-ISDN calls IF you want to operate an anlog fax on the Samsung. direct here means that the call will not travel through the Asterisk core. Side note: Stay away from solutions that use mISDN, instead go with Zaptel (DAHDI), Woomera or CAPI. Another way to go is to use an external ISDN gateway (Patton, for example), which will spare you from a lot of kernel driver update headaches in the following years. If you are planning on moving an anlog to an Asterisk-only solution on medium term, then consider to a) keep the Samsung for fax operation, or b) look at an ISDN card that has an option to sync an anlogue telephony card of the same vendor to the clock of the ISDN card (typically using a special but simple card-to-card cable). Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI vs mISDN was Re: Asterisk, VoIP and Samsung iDCS100
Hi! Side note: Stay away from solutions that use mISDN, instead go with Zaptel (DAHDI), Woomera or CAPI. Interesting. I've been usng mISDN for some years now without issues. Why should I migrate to DAHDI? None - if you are happy then don't touch it. :-) Otherwise search this list's archive. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with asterisk
Hi! [Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have 6839, digest has 3169 You most likely have two SIP UAs that use the same IP, of which the 6839 account is listed last in sip.conf while 3169 is trying to auth (unsuccessfully). Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MGCP
Hi! I have asterisk 1.4 I want to make a MGCP trunk as a client to connect to a provider who is using MGCP protocol, he provided me with user password, You will most probably need 1.8 for this, with 1.4 you will certainly not be able to succed. Read more: http://www.voip-info.org/wiki/view/Asterisk+MGCP+channels Quote Matthew Fredrickson: A new channel driver, called chan_ccs, that allows, among other things, you to control MGCP media gateways for bearer trunks, instead of having to locally terminate them on the asterisk box that's controlling the signaling links. There is also code in the same branch that has chan_ccs that modified chan_mgcp so that Asterisk can act as a media gateway (since I don't have any good real media gateways to test on). This basically means you can have Asterisk TDM channel scalability up to (in the ideal state) the same level as you can do with SIP with no media, per box. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No media being sent in SIP call
Hi! I've turned off t.38 and all of the codecs except ulaw; I still have the same problems. SOMETIMES it works. Other times, the sniffer clearly shows that the media simply isn't being sent. NOTHING is being sent. Anything else I should check? Look at the firewalls and possible SIP ALGs that are between the devices. Check for UDP port forwarding settings, and check that the RTP ports that have been negotiated for the call are not conflicting with those of other devices/calls/port forwarding settings. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto provisioning from public server
Hi! In this case I will want to use Snom phones. TFTP is available, but no FTP (with indeed then a username and password). FTP would be great... You could also consider to use the SNOM Redirection Service for provisioning: http://wiki.snom.com/PROVISIONING Remark: TR-69 provisioning doesn't appear to fit to your environment from what you have disclosed. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Channel naming conventions
Hi! but all of a sudden we have all calls origination from one sip extension opening channels which have the name of another sip extension in the channel name. Do the devices of this extension happen to have the same IP address? Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same extension registering over eth0 and eth1
Hi! Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only from eth0, and if this port fails, sends registration coming in from eth1? Spanning Tree (STP, RSTP, MSTP) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio problems on cable modem link
Hi! Can someone suggest where to look? Could this be the ITSP? - turn off IAX trunking mode - test with SIP to find if it IAX causing the trouble - capture the RTP traffice on the other side and let wireshark have a look at that stats (loss, jitter) Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio problems on cable modem link
Hi! Trunking only reduces overhead after 4+ calls, so that shouldn't help either. (Since this occurs at 2 calls) Trunking requires a timing source, and you might have trouble with your timing, that is why I suggested this (and because you did not tell us wether you have trunking enabled or not). In general IAX is not as well tested as SIP, and once in a while there are interoperability issues with IAX between Asterisk versions (and/or other IAX protocol implementations). Apart from that: you could also try a different cable modem. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone
Hi! I'm having difficulty with registering a SIP account in a Snom 320 IP- phone. Do a SIP trace on your SNOM phone, and you will most probably see that the 401 reply of Asterisk does not arrive on the phone. Then check your STUN/ICE settings on the phone in combination with the nat= settings in sip.conf on Asterisk. BTW: Are you happy with firmware 8.4.18? I still stick to 7.3.30 for the 3xx models. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intercom with Dial() works, but not with Page()
Hi! Can you tell me how I can get my Snom 320 auto-answer the call when I use the Page()-command ? Configure a special identity on the SNOM that is set to auto-answer in the phone's configuration. Or consider to use multicast instead of Page() if your network topology doesn't stand in the way of multicasting. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alert-Info advice
Hi! I've also been lead to believe that I can set a URL for a custom ringtone in the Alert-Info header but can't find the exact syntax anywhere. http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+snom http://asterisk.snom.com/index.php/Asterisk_1.4/Ringtone_Configuration Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alert-Info advice
Hi! Just out of interest, have you ever got this working? Yes, sure. Mine just isn't but I'm starting to think that my mp3 to 8000Hz Mono 16 bit wav files is a bit dodgy Very well possible. Also look at the individual identity x configuration and consider to select Custom ringtone, then enter the URL for the wav file in question. That is a good and easy way to test if that specifc wav file actually works - also check the http log and/or the snom log. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func SHARED, how to use?
Hi!
Looks like I still don't understand how SHARED works :-(
exten=6052,n,Dial(SIP/6052,,M(test))
exten=6052,n,Dial(SIP/6052,,M(test^${CHANNEL}))
Please check if in Asterisk 1.6 the Syntax for passing arguments to the M
option of Dial() has changed.
[macro-test]
exten = s,n,Set(SHARED(foo,${CHANNEL})=456 )
exten = s,n,Set(SHARED(foo,${ARG1})=456)
Philipp
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Re: [asterisk-users] tcpdump auto stats script
Hi! traffic to an IP address - then, rather than me manually analysing with wireshark, will analyze the cap file and produce stats on jitter, lag, delta etc. This is what RTCP was made for. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rtp problem with 1.8.0-rdc1
Hi! Why is it a problem? It sounds like Asterisk does silence suppression. 1) With no rtp traffic, the nat device will drop the connection in it's nat table and thus disconnecting the softphone from Asterisk. (after the router's timeout period of course) 2) The other issue is you are connected to a conference call and you want to mute your transmitter while listening to the conference. Set internaltiming to yes in asterisk.conf and see if that helps. In addition you might also be able to change the mute behaviour of your SIP clients so that it keeps on sending silent RTP packets. Philipp P.S.: You could also mute the conference user, not the SIP UA. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func SHARED, how to use?
Hi!
And the third hit in my google result is this:
http://lists.digium.com/pipermail/asterisk-users/2009-July/235288.html
Since I mentioned in my previous message that you will find the answer
in the archive of this list you could have found that even without
google. gmane.org for example has a nice web UI for reading this list.
I'm sorry, but this is absolutely the same thing I see on voip-info.org.
Indeed, I added that info to the Wiki yesterday to make it easier for you
in case you would not succed with Google.
And, I'm too stupid to understand how to use it in dial plan, especially
for RTCP statistics. :-( May be this is very-very simple, so nobody
understand what I want, if it is absolutely clear... But,could someone
provide me example of how to use SHARED with RTCP?
Since you are not describing a concrete problem in detail (and illustrate
that with some code) it is difficult to help you. For example you still
do not tell us if the Zap/DAHDI channel is your first or second call leg,
and you do not tell us which data you would like to access, and how it is
failing in your current implementation.
Anyway, usage of SHARED _is_ very simple: In the Macro that is called
with the M option of the Dial() statement you add a one-line statement of
the type
Set(SHARED(foo,SIP/123)=456)
- foo is the variable name that you choose
- 456 is the data you are interested in (e.g. the codec chosen)
- SIP/123 might be your Zap/DAHDI channel, which you could pass onto the
second call leg either by a channel variable prefixed with an underscore
(like _MYVAR), or by adding it as a Macro argument.
As a result you will be able to use
${SHARED(foo,SIP/123)}
in the h extension of your first call leg.
Note: RTCP data are only available for SIP channels, not for Zap/DADHI
channels or other channel technologies. Information about the negotiated
codec is not part of the RTCP data.
Philipp
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Re: [asterisk-users] func SHARED, how to use?
Hi!
There are 2 things I can't understand
- 1. how can I know channel name?
${CHANNEL}
2. where should I call this SHARED function? before Dial, after Dial?
Either In the macro that you specify using the M option of Dial() or in
the h extension. You will, however, have trouble treating the 2nd call
leg with the h extension, and the M option only fires at the beginning
of the call/bridge.
and this is what I tried in h:
exten = h,1,Set(CDR(userfield)=${CHANNEL(rtpqos,audio,local_count)})
This can looks different, this is just a try :-)
But Channel is DAHDI, not SIP, so I want to use SHARED.
Are you on Asterisk 1.4 or 1.6 or on 1.8? The above will not work
with 1.4.
Since I have not done this with 1.6 or 1.8: See if you can get the RTCP
data without using CHANNEL(), and instead use the individual xxxBRDIGED
RTCP channel variables as illustrated on the Wiki. Your SIP channel is
the 2nd channel (= the bridged one).
If this fails: See if AMI can help.
Philipp
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Re: [asterisk-users] func SHARED, how to use?
Hi! I see. I want to use SHARED function! Do you have example how to to export them to the local call leg/channel ? Have you considered using Google (or your favourite search engine)? The search terms asterisk function shared will surely help you, and in fact point you to the very archive of this mailing list. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func SHARED, how to use?
Hi Dmitry! Have you considered using Google (or your favourite search engine)? Shure, I searched and find nothing. The search terms C will surely help you, and in fact point you to the very archive of this mailing list. Don't know where this quote comes from, but C is absolutely not what I wrote. Instead the terms that I mentioned were: asterisk function shared And the third hit in my google result is this: http://lists.digium.com/pipermail/asterisk-users/2009-July/235288.html Since I mentioned in my previous message that you will find the answer in the archive of this list you could have found that even without google. gmane.org for example has a nice web UI for reading this list. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to open vm-INBOXs
.slin is not .wav -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a distributed paging system
Hi! I need the system to be resilient to any network partition, so that anyone can send announces from any mic to all the reachable clients. I'd need also to page a subset of all the speakers. Most of the major phone vendors (that are employed by the users of this list) have support for multi-cast of some sort. In recent firmware release notes I have read that SNOM has now also added a feature to feed multicast directly from a phone (and not just play multicast audio on the speaker as long as the phone is not in use). I'm currently using some software I wrote which sends voice over multicast RTP and coordinates all the sites with multicast messages. app_page has been around for quite some in Asterisk, and the new Asterisk 1.8 now also adds the channel driver MulticastRTP. Is there a way asterisk could be of use, or would I need to bend it too much? Look here: http://www.voip-info.org/wiki/view/Asterisk+cmd+Page http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom I have made good experience with MAST for multicasting SNOM phones: http://www.aelius.com/njh/mast/ Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func SHARED, how to use?
Hi! Could somebody tell me how to use SHARED function? http://www.voip-info.org/wiki/index.php?page=Asterisk+func+shared I want to get RTCP stats from SIP, but current channel is DAHDI. How can I get SIP channel? If you have one DADHI and one SIP channel bridged together, then only for the SIP channel you will be able to retrieve rtcp data. Depending on wether that SIP channel is the first (local) or the second (outbound or remote) call leg you will need to follow the approach described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+rtcp Quote: Use the M option of Dial() if you would like to get the codec (audionativeformat) of the remote call leg/channel and similar data. Those are not available anymore during the hangup phase (h extension), however you can store them directly in the CDR system, or use the SHARED function to export them to the local call leg/channel. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confused about notifyringing in sip.conf
Hi! notifyringing = no ; Control whether subscriptions already INUSE get sent RINGING when another call is sent (default: yes) Does this mean that when I mark this as yes, a phone that already has taken a call will be send a second and third call ?! No, not directly: This setting is only about SIP subscription information spread via NOTIFY messages, and thus only concerns the hint (BLF) feature of Asterisk. I want that if a phone is in use (calling), the phone does not ring on a second call. The term you are looking for is call waiting; use GROUP() or call-limit= to control the number of calls for a specific phone (or employ DEVICE_STATE()). Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
Hi! Does this shine new light to the problem ?! No. Once more: Go and read doc/backtrace.txt. And check if you have any meaningful information in /var/log/messages for the timestamp when asterisk crashed. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
Jonas, everyone here supports you in your effort to get a good Asterisk installation going, but could you ... maybe restrain yourself a little bit and reduce the number of hasty postings you are sending to this mailing list? Thank you, Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
Hi! I know I post a lot concerning this issue, but this is because this problem occurs on a production system and I feel very hot breathing down my neck. Why not reduce the pressure and revert to 1.4.30 for the production system until you have figued out the issue? That will give you more time for (off-hours) testing and a more thorough look at what is going on. Look in /tmp if you can't find the core file elsewhere. http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_informati on.txt Philipp P.S.: reboot usually refers to the OS, whereas Asterisk crashes or restarts. From your messages I understand that your linux box is not rebooting itself regularly. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
Hi Jonas! It indicates to be a binary file, however I have not found instructions on dealing with this @ the link you gave me. Can you give me instruction on how to handle the core.pid file ? Could I ask you again to make an effort to reduce your number of daily postings to this list? If you google for Asterisk core - or rather search this list's archive - I am quite sure you will very quickly find additional info. In case that fails (for reasons that I cannot imagine) here is another keyword to look for: gdb So please do put a little bit more effort into attacking the issues yourself before creating a post for every single tiny question that you might come across in the process. That's what everyone else here does - we all have to do our own homework first. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skip Busy Agents/Channels from Queue
Hi! DEVICE_STATE function is not available in asterisk, even DEVSTATE does not work for me in asterisk 1.4.35. Any other method function to check the channel status There is a backport available for 1.4: http://www.voip-info.org/wiki/view/Asterisk+func+device_State I assume that with does not work for me you meant that in your unpachted Asterisk 1.4 version you do not have that function available. This backport is very straigt forward to use: Just drop the file into the funcs/ directory and re-compile. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] getting error chan_sip.c: Failed to grab lock, trying again..
Hi! I am running asterisk ver 1.2.4 and have faced this error: Try a downgrade to Asterisk 0.7.1 ;- Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Macro when calling cellphone (GSM) + silence when connecting
Hi! is their a way to keep having a dialtone for the calling party when the macro is executed ?! Or not ?! Consider using either FollowMe() or using the G option of Dial() instead. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS and fixed land lines
Hi! 1. Do you have any experience with receiving incoming SMS on an analog or ISDN landline ? How can then you differentiate an SMS call from a voice call ? From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it seems the way to tell an inbound call is an SMS one is to read the callerid number but does this still apply with calls coming from cellphones ? Yes, typically there is only one SMSC that can send you SMS on a fixed line; look at its Caller ID to identify a SMS call. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS and fixed land lines
Hi! Yes, typically there is only one SMSC that can send you SMS on a fixed line; look at its Caller ID to identify a SMS call. Even when the call is coming from a cellphone ? A SMS is not really a call (at least not in the mobile world), and the cellphone cannot directly send a SMS to a landline phone. Instead it hands the SMS to the SMSC of the mobile carrier, which in turn hands it over to the SMSC of the landline carrier. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom phones recommended firmware
Hi! We're using firmware 7.3.30 on an installation of Snom 300 phones. Should we stick with it, or do the newer firmwares have better support for Asterisk? So what is it that you are missing that firmware 8 does offer? 7.3.30 is rather stable and therefore a good choice. Actually I would say this works the other way around and you could consider moving to Asterisk 1.8 soon to better match the Snom capabilites (some keywords: SRTP and TLS, P-Asserted header, AOC). That said there are interesting features that firmware 8.4.11/17 added: * support for 802.1X with eap-md5. * support for LLDP-MED (Link Layer Discovery Protocol-Media Endpoint Discovery). http://wiki.snom.com/Firmware/V8/Release_Notes/Change_Log_V8 Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6
Hi! After upgrading to Asterisk 1.6, we simply get no audio until the dialplan finishes. On the Asterisk console, I can see that the sound file is indeed playing, but we can't hear it. [...] I have tried so many things that I have lost count, and I humbly ask the collective intelligence of the Asterisk community for assistance. For a start: * upgarde to the current release of 1.6.2.x * does that message play when you call it without a forward (302) on your admin phone? * convert the .gsm prompt to a .wav or .alaw or .ulaw prompt and see if that improves matters * do a RTP debug to see if there is any RTP being sent at all * consider ChanSpy for listening in (although I doubt that'll help you) Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound SIP trunk hunting (or any fxo for that matter)
Hi! My question is this. Is it possible to tell Asterisk to execute part of a macro as a block without allowing any other commands to be processed during that time? What would be a correct way to do this in 1.4.x? *CLI show application MacroExclusive Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.0-beta4 Now Available
Hi! * Remove current STUN support from chan_sip.c. This change removes the current broken/useless STUN support from chan_sip. (Closes issue #17622. Reported by philipp2. Review: https://reviewboard.asterisk.org/r/855/) What you do not see mentioned here, and that is a bit misleading: Instead res_stun_monitor was added to be able to detect dynamic IP changes. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] parkcall: How to remove announcement.
Hi! How can I remove the Playing digits from parkcall application? In general you can address problems like this by creating your own set of sounds files where the obstructing files are either simply missing or replaced by silence. Use Set(LANGUAGE) right before the action (here: parking the call) and create your own imaginary language strucutre below /var/lib/asterisk/sounds/. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on AMD
Hi! By a mixed environment I mean some Asterisk servers running on AMD and some running on Intel If it was possible for that to matter, then the software would be very poorly written indeed. As another poster said, the only way that would have any effect is if you compiled binaries specifically for one family of processors and used them on the other. As far as how the software operates, by definition the processor type/family does not matter at all. Quite some time ago there was a difference in how the GSM codec was handled on AMD K6/Athlon systems, but that did not matter greatly, and it was just a tiny little optimisation setting in the Makefile so gain a little more speed. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial option 'r' not working anymore?
Hi! Does anyone know if the behavior of 'r' has changed but was not documented? If yes, then how does one inject ringback audio before the call is answered on the called end? Search this list for progress or progressinband, and look at the voip- info wiki. http://www.voip-info.org/wiki/view/Asterisk+cmd+Progress About 'r': There is a always the possiblity to introduce new bugs/regressions in newer Asterisk version; test two 1.4 versions and if you find a difference then consider to open a bug report. In general: The 'r' option is very seldomly what you want to use, it presents the caller ringing even if the other side is busy/congested. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: How to determine which party h angup the call? cause of Hang-up needed.
Hi! Is there ANYWAY to find out which party hang-up the call or if the call was cut-off due to other reasons? The only way - apart from putting DAHDI or SIP into debug mode - I can think of is to use the 'g' option of Dial(): If the remote side hangs up then 'g' will come into effect; if the local side hangs up this will not happen. But don't ask me what happens if Asterisk in between the two (as B2BUA man-in-the-middle) decides to to hang up... is it in this case acting on behalf of one of the two call legs? In general: It would be a great help if Asterisk provided an easy way to get at this information, i.e. who was it that hung up a call. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: How to determine which party h angup the call? cause of Hang-up needed.
read the value of var ${HANGUPCAUSE} next line to dial command.
So how can you be sure this has been populated by PRI and not by SIP?
This will not tell you which side hung up.
Philipp
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Re: [asterisk-users] Can ChanIsAvail return status from sip uri using router ip
Hi! Although my previous posts in this forum have not received satisfying answers, here is another question from me. You might want to consider to reqest a refund. ;- my question is can i use ChanIsAvail function to get the status of a user in the format SPI/user-id if i provide user in sip uri like this ChanIsAvail(SIP/u...@153.18.x.x:5062) calling user with this sip uri works fine. That will not work, you need a known peer combined with qualify= for a useful response. You can read details on the Wiki: http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)
Only when I configure my Grandstream to use only G726 (I have 8 choices), I see that the g726-codec is used. When I configure 7 x g726 and 1 x alaw, then again alaw is used ! Is it normal that Asterisk has such a great preference for alaw ?! The moment the peer suggests codec alaw (even if it is last choice), alaw is chosen by Asterisk for the communication. Please look at the first part of my last message (order of codecs in the [general] section) and apply changes there, followed by a sip reload. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outboundproxy timeout or qualify
Hi! Let's say I call by SIP/trunk1/number and the proxy server is down, is there a way to getCHANUNAVAIL? *CLI core show application Dial Unfortunatelythe timeout parameter will not do the job for me. I need somethingequivalentto qualify to monitor the outboundproxy. Why not qualify and ChanIsAvail() or SIPPEER(peername|status) if you really do not want to use the DIALSTATUS variable after your first Dial()? Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mapping of disconnect reasons
The mapping in Asterisk 1.4.24 is the problem: 402 Payment Required is mapped to 16 Normal termination instead of 21 Call Rejected. Could you direct me to the relevant file of code where these mappings are done? Before reporting a bug I would like to confirm whether this issue has been addressed on newer releases. Look in channels/chan_sip.c and search for 3398 See also: http://www.voip- info.org/wiki/index.php?page=Asterisk+variable+hangupcause Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Identify remote prompts: Partial audio matching?
Ok, here's the challenge: I would like to be able to find, match - and then react - upon prompts that are presented by the outbound/remote side of a call. Think mobile phone and This user is temporarily unavailable. Collecting a limited number of known prompt snippets should not be a problem, but how would you then detect their presence in a longer recording (or live audio stream)? Recently there was an at least slightly related posting on this list, if I recall that correctly, but I have simply not been able to turn this up. Philipp P.S.: This is all about audio analysis, not about cause codes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Identify remote prompts: Partial audio matching?
You might be able to record these snippets then pass them through the Vestec or Lumenvox Speech engine to get what you want. Unfortunately that won't work because: * the containing recordings/feeds can be quite long, can be embedded/surrounded by silence, ringing tones, music or special tones, and the ASR engines are not really designed to handle this situation. * next to this both LumenVox and Vestec do not cover the language(s) that I need this for, since both companies are focused on the American market (and yes, I am aware of Loquendo and Nuance). So that is why I am looking for something like partial audio fingerprinting; this is a bit like these find duplicate mp3 songs in my huge media library tools, only that in this case it is 1. not about an exact duplicate, and 2. the audio quality can vary, and 3. this is about finding contained parts instead of comparing full songs with each other. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mapping of disconnect reasons
Hi!
Is there a way to change the mappings of disconnect reasons to certain
SIP messages? E.G. I need to change the mapping for SIP 402 Payment
Required from 16 (normal termination) like it is in 1.4.24 to 21
(call rejected) as defined in RFC 3398.
* if you think the mapping is wrong, then you should open a ticket on the
Asterisk bug tracker
* the mapping can only be changed in the code - which you ahve
* Asterisk 1.8 will allow to read SIP response codes in the dialplan via
{HASH(SIP_CAUSE,channel-name)}. Asterisk 1.8 also comes with a
'use_q850_reason' configuration option for generating and parsing, if
available, Reason: Q.850;cause=cause code.
Philipp
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Re: [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)
Hi! Question 1 : [Aug 2 13:56:57] Capabilities: us - 0x90a (gsm|alaw|g726|g729), peer - audio=0x808 (alaw|g726)/video=0x0 (nothing), combined - 0x808 (alaw|g726) why is combined alaw|g726 and not g726|alaw (reverse) ?? Guess: Here the order presented has no meaning for the order of codec negotiation. Question 2 : why do I see on my Grandstream phone that the codec being used is alaw in stead of g726 ?? Because that is what the phone and Asterisk have negotiated. ;-) Question 3 : How can I get g726 as first preferred codec ?? Which Asterisk version are you using? * check if you have disallow/allow settings in the [general] section of sip.conf. Depending on your Asterisk version only the order in [general] would be respected, but not the order in the individual sip peer/user definition * look at the variable SIP_CODEC for the inbound (first) call leg, and in Asterisk 1.8 (or 1.6.2?) also at SIP_CODEC_OUTBOUND * many Asterisk operators have applied the third party codec negotiation patch Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)
Also: There are at least two implementations of the g726 codec, i.e. g726 and g726aal2. For this also look at the g726nonstandard setting in sip.conf. It is quite possible that your problem is here. For quick testing to see if the codec works at all: Configure your phones to do g726 only (so no alaw/ualaw at all). Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)
Hi! In the [general] section of sip.conf I have : disallow=all allow=g726 allow=alaw allow=g729 allow=gsm So change the order there and see what happens. * look at the variable SIP_CODEC for the inbound (first) call leg, and in Asterisk 1.8 (or 1.6.2?) also at SIP_CODEC_OUTBOUND When I read the value of this variable just before the Dial()-statement, it is empty. You need to set it, not read it. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 72, Issue 82
Hi! i want to get channel-id before dialing so that i can dial using that channel id. I am afraid that is not going to work. Maybe you should take a step back and describe what it actually is that you are trying to accomplish. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Please test: STUN patch for Asterisk behind NAT
Hi there! David has put up a patch to fix the STUN issues that has plagued Asterisk 1.6 ever since that feature was introduced. Now we need testers to verify the patch, so please grab the patch (or check out the SVN branch) and add your comments: https://issues.asterisk.org/view.php?id=17622 Thanks, Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR and custom name fields
Hi! I have Asterisk 1.4, trought ODBC I'm savind CDR to MSSQL. How can I change the fields name of database? You will want the adaptive CDR backport to Asterisk 1.4: https://issues.asterisk.org/view.php?id=1 http://svncommunity.digium.com/view/tilghman/branches/1.4/ Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answered call not bridged
Hi! - upgrade to a current 1.4 version, 1.4.17 is very old (you probably run this because of the zaptel -- dahdi change, but still) - do you have a SIP proxy or any SIP-aware hardware in your network that might play tricks on you, e.g. a SIP ALG (application layer gateway) on your Internet router or something similar? - enable SIP debugging on your phone and check its logs; you could also do a packet capture on your router to see what exactly is happening and if Asterisk is somehow being cut out of the loop - see if canreinvite=no somehow helps; disable STUN on your phone inside the LAN, and maybe even block direct Internet traffic for your LAN phones so that they must go through Asterisk. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing Variables From Dial Macro To Parent Ruby
Hi! Three notes: * as others have already mentioned: personally I would not Dial() from within AGI using EXEC, but rather set extension and context and then let the dialplan handle the Dial, and therefore complete that AGI before the Dial; then possibly run another AGI after the call in the h extension (even if that might not scale so well it is usually just fine). * the second call leg is already gone when call control returns to your AGI, that is why you cannot read its variables. The only way is to use your M() Macro to store the value you are interested in, either by putting it into the CDR(userfield), or by using the SHARED() function (there is a backport for Asterisk 1.4) that can export the data to the originating channel. * you need to do a GET VARIABLE in your AGI, not sure if this works flawlessly with the CHANNEL variables though Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1 second Audio Lag
Hi! I'm working for Zoiper, you can contact us directly on supp...@zoiper.com Zoa I will do a test call from a soft phone to my mobile. I can speak into my headset and the audio is heard instantly. But if I speak into my mobile there is a 1-2 second delay in the Audio. I am using SIP. I am only finding it in the Zoiper Softphones that we are using. We are able to make a call without lag on the X-lite softphone no problem. This looks very similar: https://issues.asterisk.org/view.php?id=17404 0017404: [regression] audio delay when bridging calls related to timestamp mismatch when answering an inbound call, the remote party hears a delay from 1-3 seconds. The audio is being transmitted, but the rtp timestamps take a huge jump when the call is answered even though the rtp sequencing is correct. This started occurring after 1.4.28. reproduced with 1.4.30, 1.4.32 and SVN from 05/25/2010. related to 0017007: [patch] RTP Timestamp changes after transfer, but SSRC not and the markerbit ist not set (one-way audio after transfer) Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk unresponsive
We are running asteriskNow 1.4.18 and after a few days it becomes unresponsive and inbound INVITEs timeout. Search this list for DNS. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URgent - capturing 'answered'
Hi! Great, but how exactly do i find that channel - that is my question - which command. For the third time: Use the M option to Dial() and create a Macro. In that macro use the SIPCHANINFO() or CHANNEL() function to get what you want to get. No AGI (and AGI is a protocol while Ruby is a language). If all you need to do is the the channel name of the channel that answered the phone why are you doing so much work? Version 1.4 allows for an agi to be called when the dial command is answered. Version 1.6+ allows an agi as well as a macro to be called. You can find the channel that answered a multi channel dial command. Is this not what you wanted to know? Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID disappear from CDR on transfer
Hi! 7) if john doe want to speak with caller assistant bridge the two lines using the transfer function of GXP2000 phone (REFER). After the transfer in the CDR i can't see the callerid of the caller, only data of the bridged call is reported. Any idea on what i can do to keep it ? Either store the Caller ID somewhere using your PHP script and then later add it to the CDR data of the other call, or look at at ImportVAR() or the SHARED() function (of which Tilghman has made a 1.4 backport available). In either case you will need to do some matching magic to logically link the two calls, maybe with the help of BRIDGEPEER. Possibly there are some transfer specific channel variables that can help you with that (like BLINDTRANSFER or TRANSFERSTATUS). This depends on your Asterisk version, though. Anyway, since you already determined where to transfer the call to that matching should not really be an issue. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URgent - capturing 'answered'
Hi! Depending on the version of Asterisk you are running you can call a macro or an agi as option to dial. These will be called when the line is answered and you can find the channel name of who answered. Do as he says, look at the M option to Dial. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Zap-Sip calls.
The problem we are having with Asterisk is when we initiate a call via a Zap line and it goes out on a Sip line. When it goes out via Sip we hear no sound until the party we are calling answers the line. Search for progress and/or progressinband. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vocera Comm Badges
Hi! I've worked with these before. They are designed to run a whole hospital shift, so there should be no worries regarding the battery. Sounds good. The speaker phone quality is acceptable (the speaker is quite small and points forward, not upwards in the direction of the ear), or would you rather recommend a headset? I'm not aware of the server having any kind of SIP support -- I think you would need to have a PRI trunk to another PBX. That appears to be a newly added feature. The last time I talked to them they had their own proprietary codec to deal with the occasional packet loss of Wi-Fi, and the codec was encrypted, too. Interesting - any clue on the signalling method? Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vocera Comm Badges
Hi! I´ve seen them at trade shows, I think I remember it being proprietary. What about using Dect handsets? That Star Trek device has always interested me. Too bad they chose WiFi over DECT, though. Vocera badge: * WLAN b/g * Talktime 2-2.5 hours, standby 20-27 hours * headset jack * OLED display (why don't they ever show this on the pics) * Linux based Back side: http://farm4.static.flickr.com/3412/3496101115_32319840a5.jpg System: * Windows server * Nuance ASR biometrics Dictaphone * Dialogic T1/ISDN/Analog cards * SIP interface * iPhone app I am also curious to hear some user reports, in particular about battery performance: Does it last one entire working day, or does it need to be exchanged (once?) per day? What is the smallest feasible installation? What speech protocol does the badge use to talk to the server? First quarter of 2010 was the third consecutive profitable quarter for the company. In addition, since the beginning of 2010, the company has announced the addition of 22 new employees across North America and the United Kingdom. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attended Transfer question
Hi! I've been asked to implement the following transfer workflow in an asterisk system, and I'm not seeing an easy way to do the bolded steps below (steps 4 and 5 for those with a text-only email client): You could create a dynamic meetme room for the 3 legs and drop out when done. Or do it with X static meetme rooms. You could set up a context to create the MM room and call the supervisor, connecting them to the room. Why not just do it with your phone? * Call the staff member using your 2nd line appearance * Create a phone-based 3-party conference * Dissolve the conference * Transfer the caller to the staff member If that is not what you want: Look at MeetMe() and ChanSpy(). Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_local - Asterisk 1.6.2.6
Hi! I got some reports of (Debian Testing/Unstable) systems where the timerfd timing didn't work properly and the workaround was reverting to the pthreads one. I have not yet managed to reproduce them here. I wonder if this is the issue. How about this: http://svnview.digium.com/svn/asterisk?view=revisionrevision=198146 Resolve issues with choppy sound when using res_timing_pthread. The situation that caused this problem was when continuous mode was being turned on and off while a rate was set for a timing interface. A very easy way to replicate this bug was to do a Playback() from behind a Local channel. In this scenario, a rate gets set on the channel for doing file playback. At the same time, continuous mode gets turned on and off about every 20 ms as frames get queued on to the PBX side channel from the other side of the Local channel. [...] (closes issue #14412) http://svnview.digium.com/svn/asterisk/trunk/res/res_timing_pthread.c?view =logpathrev=278465 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play alaw file with .wav extension
Hi! 2. my SOX (1.14.0) on CENTOS doesn't handle alaw files. It surely does, only that you need to tell it explicitely to: Use -t ul or -t al and you are fine. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP
Hi! I'm experiencing a problem with my SIP channel's. When I have an external connection for one of my SIP carrier's, I can listen to the client and the client listens to me normally. The problem is when I will transfer this connection, the call is mute for the extension I have transfered. Only the client hears normally. this is exactly the problem i had. Have a look at issue 17641 (https://issues.asterisk.org/view.php?id=17641) Also look here: https://issues.asterisk.org/view.php?id=17007 Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_local - Asterisk 1.6.2.6
Hi! Nobody uses chan_local Absolutely nobody. Except you. ;- Maybe this will help you: Search for Asterisk timing, consider to not run Asterisk in a virtual environment, and do not run X on the same box. Makre sure to turn off silence suppression in your SIP client(s). Search for choppy audio. Check if earlier Asterisk versions behave better. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP
Hi! client listens to me normally. The problem is when I will transfer this connection, the call is mute for the extension I have transfered. Only the client hears normally. I *think* there is/was an entry in the bug tracker on this. You might want to search https://issues.asterisk.org (also look for RTP issues with SSRC) and in the meanwhile you could reveal which version of Asterisk you are using. :) Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way audio when dialing multiple registrations
Hi! I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. You need to make sure that these two phones use *different* RTP ports, and that this is handled correctly in your router/NAT device (by port forwarding or other methods). Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk, capable of receiving from website but not from fax machine !!
Hi! 'Fax for Asterisk' is a commercial application sold by Digium. This is not their official support channel. Since you paid for the product, why not contact them directly about your problem? Maybe because having to deal with Digium support is an ... uncomfortable experience that I've made twice. It really feels like a support avoidance system, and you are an unwelcome guest that should please leave as soon as possible. I really hope that Digium takes steps to a) make their behaviour less bureaucratic when establishing a support call (try to do that on behalf of a clue-less customer of yours and you know what I mean). Also my feeling is that b) the level of competence (or familiarity with the product in question when it comes to software prodcuts) could use some improvement. If you take a look at the Digium web forum for Skype for Asterisk, for example, you will see that any kind of half-official answer or helpful reaction from Digium's side is by now non-existent - most probably due to internal policies. This just doesn't feel right, and many other companies haven proven that I can be done differently. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NAT] * + private IP + locked-down firewalls?
Hi! Provided the user doesn't have access to the firewall (eg. corporate or hotel), and the firewall doesn't allow dynamic port opening through UPnP or NAT-PMP... For those interested, I was tipped through private e-mail about using OpenVPN to open a steady tunnel between the client and Asterisk, and have the SIP client send packets through that tunnel instead of trying to connect directly. Indeed - and both Snom and Yealink offer phones with OpenVPN support built-in. That way you do not need to worry about the router/firewall. BTW: I see that the Yealink firmware now also comes with 802.1x support. Am I correct that no other popular SIP phone brand has that? Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call failed: 408 timeout
SEND 0.0.0.100:5060 ?! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logging codec used in CDR
Hi! Is there a way to log the negotiated codec that was used for each call in CDR or in a separate log file? Use CHANNEL(audionativeformat) - and do the same with the help of the M option to Dial() for the remote call leg. Store that info in the CDR userfield, or create your own field if you are on Asterisk 1.6 with the adaptive CDR columns. Note: In principle the codec can be changed in the middle of the call, however in practice this very rarely (never) happens. And while you are at it also look at RTCP stats as well. If you are still on Asterisk 1.4 then consider to apply bug/patch #10590. More details: http://www.voip-info.org/wiki/index.php?page=Asterisk+RTCP Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] p2p or p2mp for BRI
Hi! The 'M' in PtMP stands for 'Multi'. Basically PTP is the standard ISDN protocol, and PtMP is an extension of its logic to make ISDN (BRI) phones behave somewhat like analog phones: allow you to connect several of them on the same line. In other words: While you *must* have exactly one central PBX with PTP, you *can* have a central PBX or you can use BRI decentrally because PtMP is organized as a bus system. In the case of HylaFax, for example, this means that with PtMP you can run chan_capi (Asterisk) and capi4hylafax in parallel on the line so that they do not depend on each other at all (no need for IAXmodem, no fax timing issues). Numbering: With PTP you can usually obtain a number block of 2-10 consecutive numbers, while with PtMP you will probably get 2 or 3 (up to 10) individual numbers that do not have to be consecutive. At least that's the way it works in Germany and Belgium. PTP can be easily enlarged with more BRI lines, whereas with PtMP this is more difficult (due to line hunting from Telco to you, choosing an outgoing DID/MSN etc). Finally: Make sure that ISDN equipment (card, gateway) and driver supports the mode that you decide for. In the case of Asterisk chan_capi is a good choice, mISDN is not - but if you are in the US then I am have no cluee how good National ISDN support is with chan_capi. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN install on Asterisk 1.6 failing
Hi! Has anyone had experience installing it? yum install asterisk-chan_misdn I'ts the latest Trixbox Distro version and same issues exists if add in the Trixbox repo. FAILS as per below Please search this list for recent messages on mISDN, or Google it. You will find that mISDN v1 does not work with current kernels anymore, and that you would need mISDN v2, vor which howerver this (or was when I last checked) there is no chan_misdn available for Asterisk so that you would have to resort to use LCR which kind of defeats the purpose when run next Asterisk. mISDN v1 is not reliable and will give you all sorts of trouble. Look at other solutions like CAPI and chan_capi, mISDN v2, external BRI gateway, ... Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial SIP channel with no registration, timeout before CONGESTION?
When I dial a peer with no registration, SIP channel currently waits
many seconds before returning ${DIALSTATUS} CONGESTION - how can I
shorten this timeout?
Look at qualify=yes for that peer.
Use ChanIsAvail() before you dial.
Use SIPPEER(peername|status) to check registration status.
Use DB_EXISTS(SIP/Registry/peername) to check registration status.
Philipp
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Re: [asterisk-users] peer IP address in CDR
Hi! For codecs use CHANNEL function, but you will only get CallLegA codecs. Without hacking Asterisk, you will not be able to get CallLegB codecs. Patch for Asterisk 1.4.33.1 attached to get such info. Thank you! In the meanwhile I found that with the help of the M option to Dial (macro called right after connect) it is possible to access also the bridged CHANNEL variables including audionativeformat - but those variables will already be destroyed before you get to the h extension. Now I would like to find a way how to add some adaptive functionality to MySQL CDR in asterisk-addons for Asterisk 1.4 that can append self- chosen fields to the cdr table (like: codec). It appears that patch 11642 doesn't - anymore - do that successfully: https://issues.asterisk.org/view.php?id=11642 So far I do not want to switch to ODBC (Tilghman was so kind to make a backport available). Note: asterisk-addons for Asterisk 1.6 is not compatible with Asterisk 1.4. I'd also prefer to avoid to write to MySQL (or AstDB as a means to export variables from the bridged channel to the originating channel) directly from the dialplan. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Minimum modules required to run VoIP and CDR
Hi! Sometimes there is a long gap between Asterisk starting and devices being able to register. First you should check your DNS setup - it has been discussed many a times on this list. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo problem in VoIP-calls
Hi! The network setup is : analogue+GXW / softphone -- Linksys WAG160N -- Asterisk server -- ITSP -- other networks Do it step-by-step: Take the Asterisk server out of the equation, i.e. call the destination directly with your softphone or the Grandstream ATA and see if that removes the echo. That fact that both sides are hearing echo is a bit unusual - especially when calling a mobile destination things should be different. Check twice that the analog devices in the setup are ok, and replace them for a test if you can. You could also test with a destination that is run by a different operator (or is located in a different country). Another test: Use the Echo() application on Asterisk and call it from both sides. Also: You could capture the traffic and look at it with Wireshark, the delay/latency in particular. Philipp P.S.: I do think a jitter buffer matters for echo, simply because it introduces an additional delay. However the Asterisk server should not use its jitter buffer because jbforce is set to no and the Asterisk server is not the final endpoint (it only sits in between). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cmd Authenticate
Hi!
i need to save into a local variable the user's input dialed during
the cmd Authenticate(). Is there a way to do it?
Use option a of Authenticate together with ${CDR(accountcode)}
Philipp
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Re: [asterisk-users] peer IP address in CDR
Hi!
Do you already have script to capture user's IP address? If not, check
it here how I am capturing it:
http://www.ilovetovoip.com/2010/05/getting-users-ip-address-remaining-
within-the-dialplan
Or simply use one fo these:
${SIPCHANINFO(peerip)}
${SIPCHANINFO(recvip)}
${SIPCHANINFO(uri)}
More details with show function SIPCHANINFO on the CLI.
But: Anyone has an idea how to store the codec(s) that were employed for
the call in the CDR (or access it during hangup in the dialplan)?
The Wiki has a suggested patch to enhance SIPCHANINFO, but I wonder if
there is a cleaner and built-in way to do it:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sipchaninfo
Philipp
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Re: [asterisk-users] Codec negotiation
Hi! Because the codec is already chosen before the call is made, and you told it that g722 is permitted. There are all sorts of discussions in play about codec negotiation, but at this point in time, if you want different behaviour you'll need to go and code it yourself Look at the list archive - there is a codec negotiation patch around: http://lists.digium.com/pipermail/asterisk-users/2010- February/244835.html The OP might also want to consider to use different lines to the same PBX, one for normal narrowband, and another one for g722. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec negotiation
Hi! Does the 1.4.26.2-patch also work with asterisk 1.4.30 ?? Most probably - who on this list would you like to test it for you? ;- Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call file structure and syntax
Well, I¹ve tried this, and something just isn¹t right. Look here: Event: Hangup Channel: SIP/ShoreTel-1-0004 Cause: 17 Cause-txt: User busy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NO ANSWER before playback or background function?
Hi! but i want to answer the channel when dial someone and pick up the phone.not play a file. Search this list for early media and maybe also for progress. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unregister and register SIP phones by using num pad on phones?
Hi! A couple of years ago, I worked with a Alcatel IP pbx and Alcatel Sip phones, and we had the opportunity to unregister user by typing *-a number and -* again, ex * 99 *, and then the phone number/sip extension was unavailable It is entirely up to you to design the Asterisk dialplan this way; many implementations have created such a roving user one way or another. Some more complex solutions also re-provision the phones (e.g. Gemeinschaft) accordingly, while others simply make the extension available/unavailable. I am sure that the Wiki at voip-info.org has some examples. There are also phones that support hot-desking (the SNOMs do, for example), but usually I find that to be too complicated for the end user. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Local channel usage
Hi! Can anyone think of a way to play IVR after conversation initiated by Dial() terminates? You will most probably have to prevent the hangup to happen in the first place: You could, for example, join the two callers by the help of a dynamic MeetMe room, and then take action when the other parties leaves, i.e. kick the remaining user out of the room and continue in the dialplan. Here's an example for Voicemail live that uses such a technique: http://www.voip-info.org/wiki/view/Asterisk+tips+voicemail+live Another way *might* be to involve a local channel for the calling party with the /n option to prevent it from optimizing themselves away: For example: The caller's SIP channel hangs up, but the local channel that it is connected with then continues in the dialplan? Not sure if there is a way to make this work - could be that you need to twist things badly so that also the caller is in fact a callee to the local channel... Finally: Put a SIP proxy in between that catches the hangup and then takes action like a redirect (transfer). Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [AGI] What scripting language for embedded hardware?
Hey Gilles, for whatever reason your messages appear twice twice on this list. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
Hi! But why can't my phones call. The outgoing lines are PRI/DAHDI T1. No sip. No iax. Why does the asterisk machine have to resolve any address? Probably because you have one or more register = statements in your sip.conf and Asterisk is trying badly - but without success - to register itself to one or more of your providers. This then happens to block all incoming LAN SIP traffic. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to pickup an extension
Do this, for example:
exten = 1234567,1,NoOp()
exten = 1234567,n,Set(_PICKUPMARK=${EXTEN:5})
exten = 1234567,n,Dial(SIP/IPphone-1)
exten = 3456789,1,NoOp()
exten = 3456789,n,Set(_PICKUPMARK=${EXTEN:5})
exten = 3456789,n,Dial(SIP/IPphone-2)
[example-pickup]
exten = **XX,1,NoOp()
exten = **XX,n,Pickup(${EXTEN:2...@pickupmark)
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Re: [asterisk-users] Unable to pickup an extension
Hi!
exten = **XX
-- This is a local extension, a certain phone which is monitored with
BLF-lights. So if I press the button I want the phone call that made this
phone ring, not another phone.
This is NOT a local extension: It is a special local PICKUP extension
(you even named it [example-pickup]). So, in order to pickup an inbound
call that has been placed to ...67 you need to dial **67.
Small note for SNOM phones: Those can be configured to prefix a monitored
extions (67) with f.e. ** so that the phone dials **67 when pressing a
blinking BLF button.
exten = 1234567,n,Set(_PICKUPMARK=${EXTEN:5})
-- If I set the PICKUPMARK-variable the same as the DID that is called,
how can I know which DID that is set in the context [example-pickup] ??
My phone, who tries to do the pickup, creates a new channel, and the
PICKUPMARK-channel variable is of course not inherited to this newly
created channel...
I am not sure I got your question, but the phone the does the pickup does
not need to worry about inheritance of the PICKUPMARK variable - it
_sets_ that variable itself (see code for **XX) and then ask Asterisk to
search for a corresponding chanel with that value that is currently in
RINGING state.
The variable inheritance only matters for the device that is being
called, in this case IPphone-1, because that is the phone that you want
to relieve of its inbound caller.
In general: PICKUPMARK is for directed pickup where you know exactly
which call you want. Look at *8 undirected pickup (and pickup groups) if
you just want to catch whatever just started calling me.
Philipp
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Re: [asterisk-users] Unable to pickup an extension
Hi!
suppose ${SIPaccounts}=SIP/testcorp5SIP/testcorp6
exten = group,1,Set(_PICKUPMARK=${SIPaccounts})
If I was doing this I'd rather do
Set(_PICKUPMARK=group)
or
Set(_PICKUPMARK=${EXTEN})
but that is probably just me. But let's look at two of your lines:
Set(SIP/testcorp4- 0005,_PICKUPMARK=SIP/testcorp5SIP/testcorp6)
Pickup(SIP/testcorp1-0009, testco...@pickupmark)
Can you see the difference? That's what you need to change so that it
matches. SIP/testcorp5SIP/testcorp6 is not the same as testcorp6.
Philipp
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Re: [asterisk-users] Skype for Asterisk - what processors/platforms does it run on?
Hi! I understand that SfA is a binary module? There are processors it will not work on, correct? Are there limits as to operating system or distros? Requires kernel 2.6, does not work on 2.4 - as I had to find out the hard way (this is not documented - so embedded people: be aware!). Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
