Re: [asterisk-users] OT - How does the blind transfer function work on Snom-870?
Am 05.03.2015 um 15:09 schrieb James B. Byrne: On Thu, March 5, 2015 05:30, Ruben Rögels wrote: Am 05.03.2015 um 01:09 schrieb James B. Byrne: I am trying to determine how the transfer button on the Snom-870 works with Asterisk. Is the ## special code employed as when it is entered through the handset or is the blind transfer through the phone function accomplished in a different fashion? Hi, I hope I understood your question correctly. AFAIK, the transfer button sends a SIP message. Entering ## on the handset is recognized via DTMF by asterisk. I hope that I understood what I was asking for. Sometimes I do not. Yes, that is what I wanted to know. Does the implementation of the transfer button feature on the Snomp-870 use exactly the same technique as the ## feature code entered through the dial pad and produce exactly the same SIP message that Asterisk produces when it gets the ## DTMF? The reason is that I wish to be able to detect a call transfer performed via either method (## or Transfer-Button) and process the result of both in the same fashion. If the button and DTMF transfers are not performed using the same switching techniques in Asterisk then I need to discover what those differences are. If both are totally equivalent from a SIP message signalling point of view then the issue is far easier to handle. I searched, in vain, in the Snom-870 docs for specifics on this and either could not find or did not recognize anything that applied. Do you know where I can locate these sorts of details. My knowledge of SIP/RTP/VOIP etc. is cursory but, given an adequate reference, I can usually sort things out. Hi again, I'm glad to hear that I provided a somehow useful answer. Unfortunatelly, I don't know these details. If you wasn't lucky consulting the snom docs, maybe the snom support can be helpful with information about the exact implementation details. You also could use sip debug on asterisk to check what's going on when pressing the transfer button vs. what's happening when using ## via DTMF. Are you forced to get the transfer information from the SIP signaling, or can you use AMI events for example? I think this would be possible if asterisk is configured to stay in the media path, so re-inviting is handled over asterisk itself and therefore detectable with AMI events. Regards, Ruben -- _ -- 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] OT - How does the blind transfer function work on Snom-870?
Am 05.03.2015 um 01:09 schrieb James B. Byrne: I am trying to determine how the transfer button on the Snom-870 works with Asterisk. Is the ## special code employed as when it is entered through the handset or is the blind transfer through the phone function accomplished in a different fashion? Hi, I hope I understood your question correctly. AFAIK, the transfer button sends a SIP message. Entering ## on the handset is recognized via DTMF by asterisk. -- _ -- 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] Intruder
Hi Felix, you have several things to check: netstat -a -n --udp --tcp will show you connections and connection attempts on network layer level. You have to look for incoming connections to port 5060 and if the call has been established for connections on your rtp ports. (see rtp.conf). If you can see connections not supposed to be there: thats your intruder ;-) I suggest you disable guest calls and you configure a default context in which dialed extensions can't be routed to charged destinations. sip.conf: allowguests=no defaultcontext=default extensions.conf: [default] exten = _X.,1,Answer() exten = _X.,n,PlayBack(silence/1) exten = _X.,n,PlayBack(ss-noservice) exten = _X.,n,PlayBack(silence/1) exten = _X.,n,MusicOnHold(default,10) exten = _X.,n,PlayBack(silence/1) exten = _X.,n,PlayBack(vm-goodbye) exten = _X.,n,HangUp() The next step would be using fail2ban or something similiar to check the asterisk log for intruders. fail2ban recognized them and dynamically sets appropriate firewall rules. Good luck. best regards, Ruben Am 16.11.2012 17:20, schrieb Felix Vazquez: I am in the asterisk CLI and can see an unidentified caller trying the make calls out of the asterisk system. How do I stop them? How do I identify them and how can I see how the go in? This is an example of what I would see: NOTICE[4098]: chan_sip.c:20063 handle_request_invite: Call *from '' *to extension '90111235551212' rejected because extension not found. Felix This electronic message contains information from BOSH Global Services which may be company sensitive, proprietary, privileged or otherwise protected from disclosure. The information is intended to be used solely by the recipient(s) named above. If you are not an intended recipient, be aware that any review, disclosure, copying, distribution or use of this transmission or its contents is prohibited. If you have received this transmission in error, please notify the sender immediately. -- _ -- 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 -- _ -- 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] Installation Problem with asterisk 1.6
Hi Akhilesh, I got below error: configure: *** XML documentation will not be available because the 'libxml2' development package is missing. configure: *** Please run the 'configure' script with the '--disable-xmldoc' parameter option configure: *** or install the 'libxml2' development package. why don't you just do ./configure --disable-xmldoc Do you really need the XML doc? regards, Ruben -- _ -- 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] Check for the voicemail
just another thought: if you send the message by mail, do you need to save it? regards, Ruben Am 21.08.2012 18:45, schrieb Danny Nicholas: Assuming that you are using the standard 100 message limit, just check for INBOX/MSG0100.txt and send the message. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danilo Dionisi Sent: Tuesday, August 21, 2012 11:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Check for the voicemail Hi all, I have a problem with voicemail. My boss has asked me to send via email, the message that a user leaves on the voicemail. This is very easy. :) After, he asked me to check before sending the email, if the receiver's mailbox is full. If the mailbox is full, Asterisk should call the receveir intern (example 2001) and using a Playback tell him that his mailbox is full. How can I do? :( Danilo -- _ -- 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 -- _ -- 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 -- _ -- 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] Check for the voicemail
Okay, so have a look at mailcmd= option in voicemail.conf mailbox will mean a e-mail-box in the next lines. What you need to do is wirting a shell script or what ever to check for the return code of the smtp session (normally it should be a 450 in case of full mailbox). In case of 450 mailbox full whatsoever you need to create a call file in /var/spool/asterisk/outgoing directing your recipient phone to a special extension which handles the playback of the your mailbox is full message. But: this will only work if you deliver the mail directly to the recipients mail server, if you use a smart host, it will accept the message and the final recipient server will bounce the mail back to you. I hope this helps. regards, Ruben Am 21.08.2012 18:52, schrieb Danilo Dionisi: I'm sorry, I haven't been clear. I do not have to check the inbox on Asterisk, but I have to check the free space on a particular mailbox of Exchange software. It's possible with the pair Asterisk-Sendmail? Il 21/08/12 18:45, Danny Nicholas ha scritto: Assuming that you are using the standard 100 message limit, just check for INBOX/MSG0100.txt and send the message. -- _ -- 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] Check for the voicemail
Hello, no problem at all, I think this is the tricky part. A smtp dialogue between your email client and a smtp server normally looks like this: user@box:~? netcat mx1.example.com 220 postfix ESMTP mx1.example.com helo me.local 250 mx1.example.com mail from: ruben.roeg...@wiseape.de 250 2.1.0 Ok rcpt to: ruben.roeg...@example.com 450 5.7.1 ruben.roeg...@example.com: Mailbox Full The tricky part is writing or finding a console smtp client that gives you feedback about the 450 error that just happened. Right now I cannot give you a precise way to do that, but I have basic understanding of the technology, so I know that it is possible to do so ;-) I'm looking around in the net, because I think I'll soon have to handle your problem aswell in my company ;-) If I can find solution, I'll post it. regards, Ruben Am 21.08.2012 19:20, schrieb Danilo Dionisi: I'm sorry, how i can to check for the return code of the smtp session? I've never done :p Thanks, Danilo Il 21/08/12 19:05, Ruben Rögels ha scritto: Okay, so have a look at mailcmd= option in voicemail.conf mailbox will mean a e-mail-box in the next lines. What you need to do is wirting a shell script or what ever to check for the return code of the smtp session (normally it should be a 450 in case of full mailbox). In case of 450 mailbox full whatsoever you need to create a call file in /var/spool/asterisk/outgoing directing your recipient phone to a special extension which handles the playback of the your mailbox is full message. But: this will only work if you deliver the mail directly to the recipients mail server, if you use a smart host, it will accept the message and the final recipient server will bounce the mail back to you. I hope this helps. regards, Ruben Am 21.08.2012 18:52, schrieb Danilo Dionisi: I'm sorry, I haven't been clear. I do not have to check the inbox on Asterisk, but I have to check the free space on a particular mailbox of Exchange software. It's possible with the pair Asterisk-Sendmail? Il 21/08/12 18:45, Danny Nicholas ha scritto: Assuming that you are using the standard 100 message limit, just check for INBOX/MSG0100.txt and send the message. -- _ -- 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 -- _ -- 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 Dahdi 1.6.2.23 Iaxmodem
Am 01.08.2012 17:15, schrieb motty.cruz: Hello, I have anolog lines coming throug Dahdi to Asterisk Server, one of the anolog lines is used for fax line. I received fax fine without any problems using Iaxmodem with Hylafax Server. Outgoint fax is the problem, when IAXMODEM dial out using Dahdi channel, dahdi answers and start to dial the outside number however Iaxmodem thinks that dahdi is the remote fax machine and starts sending fax data eventually giving up. Here is my dialplan in extensions.conf [fax-out] exten = _XXX,1,Set(__SIP_CODEC=alaw) exten = _XXX,2,Dial(dahdi/g3) If the remote machine answers within the first ring, the outgoing fax works fine, but if remote fax machine does not answer within the first ring, I get the fallowing erro: no carrier found Hylfafax and Iaxmodem are working fine, my problem is between iaxmodem dialing and though dahdi channel. Thanks, Hi, I think you should configure hylafax to listen some more on the line for a fax carrier. I'm not a hylafax expert, but there should be a configuration parameter how long hylafax should watch the line fo a fax carrier. best regards, Ruben -- _ -- 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] Garbled voicemail
Hi Dan, my wild speculation: It's some kind of timing/synchronisation problem. Do you use jitter buffer an/or echo cancelation? Best regards, Ruben -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Dan Ritter Gesendet: Donnerstag, 9. Februar 2012 17:33 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] Garbled voicemail Our Asterisk system (1.8.8.1-1digium1~squeeze) has been very stable and generally doing a good job -- except that one day, voicemail recordings started being garbled. It only manifests when the VM comes from our telco gateway service -- OnSIP/Junction -- and not from internal phones or from an Asterisk box I have at home. We have voicemail set to record to WAV, and real files are being generated -- but it sounds incredibly sped up, faster than chipmunks. Completely unintelligible, even if you pull it into an audio editor and slow down playback. It is not perfectly consistent, but it happens in about 85% of voicemail recordings left from the outside world through OnSIP. We've had several years of trouble-free voicemail before this. Anyone seen anything similar? Advice? Wild speculation? -dsr- -- _ -- 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 -- _ -- 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] asterisk not connecting to sipgate / NAT related issue?
Hello, I configured asterisk in sip.conf like that: = register = username:sec...@sipgate.de:5060/number [sipgate-out] port=5060 type=friend insecure=invite nat=yes username=username fromuser=username fromdomain=sipgate.de secret=secret host=sipgate.de qualify=5000 canreinvite=no = But all I get on CLI is: [Jan 19 15:59:55] NOTICE[1928]: chan_sip.c:12104 sip_reg_timeout:-- Registration for 'usern...@sipgate.de' timed out, trying again (Attempt #1) My first impression was that there must be a NAT related issue, so I captured some SIP debug packets and got: = Retransmitting #6 (NAT) to 217.10.79.9:5060: OPTIONS sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 192.168.23.30:5060;branch=z9hG4bK4c5e9a7f;rport Max-Forwards: 70 From: asterisk sip:asterisk@192.168.23.30;tag=as669e138a To: sip:sipgate.de Contact: sip:asterisk@192.168.23.30 Call-ID: 383e3671142f9ec4312d5a263b225629@192.168.23.30 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.21 Date: Thu, 19 Jan 2012 15:00:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 = As you can see, asterisk writes the local ip address in the VIA header. Is this the supposed behaviour? As far as my understanding of the NAT problem goes, this is exactly what I don't want to be happening. I'm familiar with the general NAT problem with layer 7 protocols relying on ip addresses in the content of a layer 3 packet, but I thought, asterisk would replace the local ip with the WAN ip. Even setting externip=xx.xx.xx.xx does not change anything. Anyone can give me advice? kind regards, Ruben -- _ -- 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 check currently used libraries from command line ?
Hi Olivier, I suppose you give strace a try. It's a powerful debugging utility, you should be able to find everything you are looking for. best regards, Ruben Am 16.01.2012 11:14, schrieb Olivier: Hi, I've recently upgraded a system from 1.8 to asterisk 10 and also updated spandsp while doing so. I wondered what is the safest and easiest way to check from command line which libraries a running Asterisk system is currently using (just like dahdi show version, for instance). Though I'm currently asking this for spandsp, this question is on a more general plan (for example, which ssl library am I currently using ?). Suggestions ? Regards -- _ -- 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 -- _ -- 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 Installation in Asterisk
Am 12.01.2012 18:50, schrieb mahesh katta: I was search for free license but for this Digium require purchase any Hardware then they can provide Free License. But I have no Digium Device , I am using Grand stream FXO Gateway and Asterisk.1.8.XX . I was connected like PSTN==FXOGateway==Asterisk(FXO configure through IP) If anything wrong please correct me. Hi Mahesh, the FreeFax for asterisk is really free and not bound to digium hardware, but it is limited to one concurrent fax session. At least you should be able to try if fax receiving is possible with this setup. As far as I can see, it should work with your setup. The URL I posted leads you to the FreeFAX for Asterisk Module. best regards, Ruben -- _ -- 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 Installation in Asterisk
Am 12.01.2012 12:44, schrieb mahesh katta:
Hi,
Any one give me about FAX in Asterisk.
PSTNFXO GATEWAYASTERISK-1.4.27(OR)ASTERISK-1.8.X.X
whenever some one is Fax to PSTN its convert into pdf format
Help me any links or pdf .. for setup this. ?
Best Regards,
Mahesh Katta
**
Hi Mahesh,
this is my macro in asterisk to handle fax:
[macro-faxin]
; Faxe
; ARG1 = eMail-Adresse
exten = s,1,Verbose(${BOUNDARY} Eingehender Ruf von ${CALLERID(num)})
exten = s,n,Verbose(${BOUNDARY} BCHANNELINFO ${BCHANNELINFO})
; nur verarbeiten, wenn B-Kanal frei ist
exten = s,n,GotoIf($[${BCHANNELINFO} = 2]?hangup:free)
exten = s,n(free),NoOp()
exten = s,n,Set(TO=${ARG1})
exten = s,n,Set(EXT=${MACRO_EXTEN})
exten = s,n,Verbose(1,${BOUNDARY} Eingehendes Fax ${CDR(uniqueid)})
exten = s,n,Set(FAXFILE=/tmp/fax-${TO}-${CDR(uniqueid)}.tif)
exten = s,n,Set(LOCALSTATIONID=jumping frog)
exten = s,n,Answer()
exten = s,n,Wait(3)
exten = s,n,ReceiveFAX(${FAXFILE},d)
This is an ugly work-around to handle fax properly becaus I can't catch
the hang-up event by the macro itself:
;fax oder kein fax, das ist hier die Frage...
exten = h,1,Verbose(${BOUNDARY} ${EXT})
exten = h,n,System(/usr/local/bin/fax2mail.sh ${FAXFILE} ${TO})
And this is the bash script to convert tif to pdf and send it via email
to my users:
#!/bin/bash
FAXFILE=$1
RECIPIENT=$2
SUBJECT=[Fax] Sie haben ein Fax erhalten
BODYSUCCESS=/usr/local/bin/bodysuccess.txt
BODYFAILED=/usr/local/bin/bodyfailed.txt
PDF=/tmp/fax-`date +%s`.pdf
tiff2pdf $FAXFILE $PDF
# Konvertierung okay?
if [ $? == 0 ]; then
mutt -s $SUBJECT -a $PDF -- $RECIPIENT $BODYSUCCESS
# Hats geklappt?
if [ $? == 0 ]; then
exit 0
else
exit 1
fi
else
mutt -s $RECIPIENT $BODYFAILED
exit 1
fi
I hope this helps!
best regards,
Ruben
--
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Re: [asterisk-users] FAX Installation in Asterisk
Am 12.01.2012 14:09, schrieb mahesh katta: WARNING[6982]: pbx.c:1851 pbx_extension_helper: No application 'ReceiveFAX' for extension (macro-faxin, s, 12) [Jan 12 18:36:00] == Spawn extension (macro-faxin, s, 12) exited non-zero on 'SIP/gxw-000b' Best Regards, Mahesh Katta *BUZZ**WORKS*Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD |MUMBAI| DELHI 222, Arunvihar,Sector-28, Noida 201301 GSM+91.3 45699 | Phone +91.12.0431.0581 Webhttp://www.buzzworks.com http://www.buzzworks.com/ Hi Mahesh, I'm running asterisk 1.6.2.21 on Ubuntu 10.4.3 LTS On a Debian based linux distribution you need the following packets: libtiff-tools libtiff4 To receive fax, you'll need the ReceiveFAX Application. You can get it for personal use directly from digium.com , just search for FreeFAX for asterisk (http://store.digium.com/productview.php?product_code=804-7) best regards, Ruben -- _ -- 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 find out one way latency
Am 30.11.2011 21:47, schrieb NaJIm: Hi All, How can I find out One way latency from my PBX to my SIP Trunk Provider. My SIP provider recommends a One way latency of 100ms for good Voice quality. Ping request to their IP Address gives me a response in approx. 260ms. Will that be good enough for a SIP Trunk. Please help. We are trying to sign up with a new SIP Provider. Thanks, Najim -- _ -- 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 Hi Najim, a ping is the time a packet needs for travelling to a destination and back to you. So the one way latency you are refering to, should be half the time your ping took. In your case this will be 130ms, I would say this is still reasonable. regards, Ruben -- _ -- 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] OT: Does IEEE 801.2q include VLAN trunking?
Hello List, I'm a little bit confused as I read about IEEE 801.2.q So, my actual question is: Does a switch stating to support IEEE 801.2q also supports VLAN trunking? As I understand the standard, I suppose it does, but I'm not sure. Can someone clarify this for me, please? Thank you vermy much. Best regards, Ruben -- _ -- 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] OT: Does IEEE 801.2q include VLAN trunking?
On 29.11.2011 11:45, Doug Lytle wrote: Ruben Rögels wrote: Does a switch stating to support IEEE 801.2q also supports VLAN trunking? I don't know if you miss-typed or not. But, 802.1q is VLAN. If it was typed incorrectly, then yes. I just recently setup a Linux DHCP to handle multiple networks with VLAN 802.1q trunking. Doug Hi Doug, thank you very much for your response, it is very helpful to me. Yes, I misstyped it, it was supposed to be IEEE 802.1q regards, Ruben -- _ -- 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] OT: Does IEEE 801.2q include VLAN trunking?
Am 29.11.2011 14:41, schrieb Douglas Mortensen: Yes. That's exactly what 802.1q is. Technically 802.1q allows the network devices to tag each Ethernet frame with a VLAN ID. This way if you have 3 vlans, they can all be trunked over 1 Ethernet port by means of tagging the VLAN ID. - Doug Mortensen Sent via DroidX2 on Verizon Wireless™ Hi Doug, ok - thank you. I also read on http://en.wikipedia.org/wiki/IEEE_802.1q#Trunk_ports_and_the_native_VLAN ---Quote start--- Not all vendors use the concept of trunk ports and native VLANs. Annex D to the 1998 802.1Q standard uses the concept of trunk links, but the current (IEEE Std 802.1D- 2004) standard does not use the terms trunk or native. Some use the term Qtrunk to avoid confusion with 802.3ad link aggregation that is often named a trunk as well. ---Quote end--- Confusing. I'm planning to use a HP ProCurve V1810G-24 or ProCurve E2510G-24 together with a MikroTik RB450 or RB1100 (which also supports IEEE 802.1q). We want to set up routed VLANs for VoIP an company's LAN without the need of using multiple ports on the router. As I understand you, this will be possible without any problems, because tagged VLANS also includes VLAN trunking, right? Thank you very much. regards, Ruben -- _ -- 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 build sip pvt data - Switching equipment congestion
Number of wished concurrent calls times 4 = Number of ports you'll need to setup in rtp.conf ;-) regards, Ruben Am 02.11.2011 16:05, schrieb Jonas Kellens: Hello, thank you for your answer... Current range (rtp.conf) : 11500 - 11650 Current calls : 20 à 25 Is this not sufficient ?? Jonas. On 11/02/2011 04:00 PM, Danny Nicholas wrote: You have set an insufficient range in rtp.conf. Asterisk uses 2 ports per call, but allocates 4 for transferring, etc, so when you set up a range of 10001-10040 (for example) you are basically setting a range of 10 concurrent calls. Check rtp.conf and make the end range larger by 8 or 12 or whatever number of extra calls you’d like to see before you get this message again. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, November 02, 2011 9:57 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Unable to build sip pvt data - Switching equipment congestion Hello list, can anyone tell me what the following means (found in messages log) : /[Nov 2 11:16:21] ERROR[18407] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Nov 2 11:16:21] WARNING[18407] chan_sip.c: Unable to create RTP audio session: Address already in use [Nov 2 11:16:21] ERROR[18407] chan_sip.c: Unable to build sip pvt data for 'sipaccount7' (Out of memory or socket error) [Nov 2 11:16:21] WARNING[18407] app_dial.c: Unable to create channel of type 'SIP' (cause 42 - Switching equipment congestion)/ Thank your for explaining the problems and a possible solution ! Greetingz, Jonas. -- _ -- 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 -- _ -- 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 -- _ -- 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.conf and Asterisk as a sip agent/client
Am 04.10.2011 10:33, schrieb Sebastian Arcus: Hello list, I use Asterisk with one sipgate.co.uk trunk. Asterisk connects to sipgate.co.uk as a sip agent/client (with register = statement in sip.conf). If I restrict the number of ports used in rtp.conf (to 1-10005 for example) - will that affect the sip sessions to sipgate.co.uk as well - or only those sessions where Asterisk acts as a sip proxy/server? Many thanks, Sebastian Hi Sebastian, it also affects the sip/rtp sessions to sipgate.co.uk. best regards, Ruben -- _ -- 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] Force a SIP friend to use a certain IP?
So I was hoping I would be able to set the source IP that we use when talking to the two different SIP friends. I see externip in general options, but is there nothing equivalent that can be set per user/peer? Hi, as far as I know, you cant do this on a per peer basis. I suppose you run two asterisk daemons, each one of them on a different external IP. In this setup you can route calls from A over one asterisk daemon and calls from B over the other asterisk daemon. Sounds a little bit like an overkill scenario, but it woul work. best regards, Ruben -- _ -- 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] SRV question
Am 08.08.2011 18:37, schrieb J Gao: Hello, All, I have a question about using SRV record. One of SIP provider is using DNS SRV record. If I use IP address of the SIP proxy server I can successfully register my Asterisk 1.8.5. But If I try to use the domain name like: /register = user:p...@somedomain.com/ then the registration failed. So I have to /dig somedomain.com SRV/ to find out the SIP proxy host, for example, sip.regserver.com, then I have to run /dig sip.regserver.com/ to find out the real IP address. Does Asterisk has the feature so I can register by SRV record? I lookup up sip.conf and I enabled srvlookup=yes, but that is only for SIP URI outgoing call, not for SIP registration. Thank you in advance. Jian Did you set srvlookup = yes in your sip.conf? regards, Ruben -- _ -- 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] ulimit
is it possible to prevent 100% cpu usage by asterisk, with ulimit? Hi, it is possible, but not recommended. There is a reason, why the asterisk process needs 100% of CPU. What is your scenario? How many extension? What type of extensions? (SIP, etc) How many concurrent calls? Hardware? You are trying to cure the symptom, not the cause ;-) regards, Ruben -- _ -- 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] Answering machine answers after pickup a phone.
Hi! I'm sorry that I have misundertood your question, didn't read it carefully enough. So you have your asterisk and your phone conntected to the same incoming line. Maybe you can try with to detect an answered call with BackGroundDetect() exten = s,1,Answer() exten = s,n,BackGroundDetect(silence/10) exten = s,n,Voicemail(1234) exten = talk,1,HangUp() I can't try it for your setup with a POTS line, but I think this might work, especially when you tune the time values for BackGroundDetect(). Quote of the manual: --- SNIP --- -= Info about application 'BackgroundDetect' =- [Synopsis] Background a file with talk detect [Description] BackgroundDetect(filename[|sil[|min|[max]]]): Plays back a given filename, waiting for interruption from a given digit (the digit must start the beginning of a valid extension, or it will be ignored). During the playback of the file, audio is monitored in the receive direction, and if a period of non-silence which is greater than 'min' ms yet less than 'max' ms is followed by silence for at least 'sil' ms then the audio playback is aborted and processing jumps to the 'talk' extension if available. If unspecified, sil, min, and max default to 1000, 100, and infinity respectively. --- SNAP --- Hope this helps. regards, Ruben Am 05.08.2011 10:59, schrieb Jorge Barreiro: Hi again, thanks for your answer, but it didn't solve the problem. That Dial command returns inmediately, so I don't even have the delay. I'll try to explain myself better. The PBX has only one FXO card, connected to the PSTN. There is no other phones connected to the PBX nor SIP extensions. There are analog phones connected to the same PSTN. What I try to do is that, when there is an incoming call from the ouside, if someone answers on a phone, then the PBX won't answer. Thanks. O Venres, 5 de Agosto de 2011 00:04:02 Ruben Rögels escribiu: Hi, your concept using Wait() won't work here. Try it like this: [incoming] exten = s,1,Dial(DAHDI/1234,30) ; This will ring the phone 30s exten = s,n,BackGround(wellcome-message) exten = s,n,Voicemail(1234) exten = #,1,Hangup() So, of you answer the call within 30s, you'll get the call on your phone. After 30s, the Voicemail will answer the phone. regards, Ruben Am 04.08.2011 21:39, schrieb Jorge Barreiro: Hello, I'm configuring an Asterisk PBX to use as an answering machine. I have a FXO card connected to the line, and other analog telephones connected to the same line. The PBX answers and redirects you to the voicemail after a delay. The problem is that even if I pickup any analog phone in the line before the PBX does, it answers after the delay anyway. And I couldn't find how to prevent this, or even if this is supposed to happen. My FXO card is a cheap X100P (source of problems, I know), and I'm using the Asterisk version included in Debian Squeeze (1.6.2.9). My dial plan looks like this: [incoming] exten = s,1,Wait(8) exten = s,2,Answer exten = s,3,BackGround(wellcome-message) exten = s,4,Voicemail(1234) exten = #,1,Hangup I don't know if this is related, but I'm in Spain and I had to add: hanguponpolarityswitch=yes to the chan_dahdi.conf file so that asterisk detects the remote hangup. I also added: answeronpolarityswitch=yes but this didn't help. It seems to be used just to detect the answer when you are calling, not when receiving a call. I'd appreciate any help you could provide. Thanks! -- _ -- 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 -- _ -- 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 -- jumping frog - Webhosting Housing Ruben Rögels Moltkestraße 24 79098 Freiburg Tel.: 0761 / 384 78 85 Web: http://www.jumping-frog.org/ eMail: ruben.roeg...@jumping-frog.org Support: http://support.jumping-frog.org -- _ -- 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] Answering machine answers after pickup a phone.
Hi, your concept using Wait() won't work here. Try it like this: [incoming] exten = s,1,Dial(DAHDI/1234,30) ; This will ring the phone 30s exten = s,n,BackGround(wellcome-message) exten = s,n,Voicemail(1234) exten = #,1,Hangup() So, of you answer the call within 30s, you'll get the call on your phone. After 30s, the Voicemail will answer the phone. regards, Ruben Am 04.08.2011 21:39, schrieb Jorge Barreiro: Hello, I'm configuring an Asterisk PBX to use as an answering machine. I have a FXO card connected to the line, and other analog telephones connected to the same line. The PBX answers and redirects you to the voicemail after a delay. The problem is that even if I pickup any analog phone in the line before the PBX does, it answers after the delay anyway. And I couldn't find how to prevent this, or even if this is supposed to happen. My FXO card is a cheap X100P (source of problems, I know), and I'm using the Asterisk version included in Debian Squeeze (1.6.2.9). My dial plan looks like this: [incoming] exten = s,1,Wait(8) exten = s,2,Answer exten = s,3,BackGround(wellcome-message) exten = s,4,Voicemail(1234) exten = #,1,Hangup I don't know if this is related, but I'm in Spain and I had to add: hanguponpolarityswitch=yes to the chan_dahdi.conf file so that asterisk detects the remote hangup. I also added: answeronpolarityswitch=yes but this didn't help. It seems to be used just to detect the answer when you are calling, not when receiving a call. I'd appreciate any help you could provide. Thanks! -- _ -- 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 -- _ -- 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 no traction for Windows version?
...and why do we all mess around with IT stuff and asterisk in special? Spoiler: because we can...! ;-) regards, Ruben Am 26.07.2011 10:16, schrieb A J Stiles: On Tuesday 26 Jul 2011, Gilles wrote: On Tue, 26 Jul 2011 07:28:27 +, Soeren Malchow (MCon) soeren.malc...@mcon.net wrote: And asterisk just runs fine on linux why bother ? Because I, for one, would like to run Asterisk on my Windows workstation at home as an enhanced answering machine :-) And you can't just run Asterisk on a separate Linux box at home as an enhanced answering machine because . ? -- _ -- 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 in Detecting Dtmf on FXO line.
Am 08.07.2011 08:58, schrieb DHAVAL INDRODIYA: Hi All, I am having Problem in detecting DTMF on analog lines. basically are system is in india and telco provider is BSNL [Bharat sanchar Nigam LImited]. We have Purchased Analog card From chinaroby.com http://chinaroby.com which is X1600P 16 port FXO card. they also provide us wctdm.c file. card is detected successfully, incoming and outgoing calls scenario is also fine. we are unable to receive dtmf properly it means there is some digit are missing when we receive dtmf the ratio of sucess is about to 70% and 30% of calls are getting wrong dtmf . Dahdi version is 2.3.0.1 and asterisk version 1.6.0.24 I load module using modprobe wctdm opermod=INDIA cidbeforering=1 cidbuflen=1 fixedtimepolarity=16 here id chan_dahdi.conf. Hello, did you try plaing with rxgain and txgain? When I set up a TDM400, I had some issues with DTMF because the signals where overmodulated. Regards, Ruben -- _ -- 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] Agents outbound calls to be recorded
Hi All; I know that incoming calls for the agent can be recorded, but how I can let the outbound calls for the agents to be recorded? I can determine the directory to store the outbound calls of the agents to be other than the directory to store the incoming calls of the agents? Regards Bilal Hi Bilal, use MixMonitor() in the outgoing extension for your agents. Regards, Ruben -- _ -- 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] SendFax: not setting the fax header
Hello,
after I solved my problem with the fax processing after receiving,
I got another problem while sending a fax: the header is not set properly.
I use a PHP_Script to upload a PDF file and to generate a call file.
A bash script is looking for existent call files in the web directory
and moves
them the asterisk's outgoing directory.
Ok, my call file looks like this:
==
$cf_commands = Channel: local/$ext@common\n
.MaxRetries: 5\n
.RetryTime: 60\n
.WaitTime: 60\n
.Context: fax\n
.Extension: 100\n
.Set: FAXFILE=$file_dst\n
.Set: FAXOPT(localstationid)=$_POST[mynumber]\n
.Set: FAXOPT(headerinfo)=$_POST[mynumber]
$_POST[myname] $_POST[myemail]\n
.;
==
As you can see, I set the LOCALHEADERINFO to string I got from a web form.
My fax extension for sending fax looks like that:
==
[fax]
exten = 100,1,System(/usr/local/bin/pdf2fax.sh ${FAXFILE})
exten = 100,n,Wait(1)
exten = 100,n,Verbose(${BOUNDARY} ${FAXOPT(localstationid)}
${FAXOPT(headerinfo)})
exten = 100,n,Verbose(${BOUNDARY} ${FAXFILE})
exten = 100,n,SendFAX(${FAXFILE}converted)
exten = 100,n,Verbose(1,### FAXSTATUS: ${FAXSTATUS})
exten = 100,n,Verbose(1,###FAXERROR: ${FAXERROR})
exten = 100,n,Verbose(1,### FAXMODE: ${FAXMODE})
exten = 100,n,Verbose(1,###FAXPAGES: ${FAXPAGES})
exten = 100,n,Verbose(1,### FAXBITRATE: ${FAXBITRATE})
exten = 100,n,Verbose(1,### FAXRESOLUTION: ${FAXRESOLUTION})
exten = 100,n,Verbose(1,### REMOTESTATIONID: ${REMOTESTATIONID})
exten = 100,n,HangUp()
==
The Verbose() application shows to me that ${FAXOPT(headerinfo)} is
correctly set,
but the headerinfo is not appearing on the fax itself...
It's not important if I set FAXOPT(headerinfo) or LOCALHEADERINFO, the
result is always the same: no header on the fax received.
I even tried to set FAXOPT(headerinfo) directly in the fax extension:
same result: no header on the fax...
Console output:
==
-- Executing [100@fax:3] Verbose(CAPI/ISDN1#02/12345-b,
### Ruben Roegels ruben.roeg...@jumping-frog.org) in new stack
### Ruben Roegels ruben.roeg...@jumping-frog.org
-- Executing [100@fax:4] Verbose(CAPI/ISDN1#02/12345-b,
###
/var/www/webfax/out/13094318919d62dbef2f0516c87cba828a4e52b6267537f313)
-- Executing [100@fax:5] SendFAX(CAPI/ISDN1#02/12345-b,
/var/www/webfax/out/13094318919d62dbef2f0516c87cba828a4e52b6267537f313converted)
in new stack
==
Again: Anyone seeing what I'm missing?
Thank you very much.
Regards,
Ruben
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[asterisk-users] dialplan execution stops after ReceiveFax
Hello,
I have a noticed strange behavior in Asterisk 1.6.18.2 with ReceiveFax
Digium FAX Driver: 1.6.2.0_1.3.0 (optimized for i686_32).
I use a context [capi-in] for icoming ISDN calls:
==
[capi-in]
; Faxe fuer Ruben
exten = 12345,1,Macro(faxin,ruben.roeg...@jumping-frog.org,${EXTEN})
==
My macro for the fax receiving looks like that:
==
[macro-faxin]
; Faxe
; ARG1 = eMail-Adresse
exten = s,1,Verbose(${BOUNDARY} Eingehender Ruf von ${CALLERID(num)})
exten = s,n,Verbose(${BOUNDARY} BCHANNELINFO ${BCHANNELINFO})
; nur verarbeiten, wenn B-Kanal frei ist
exten = s,n,GotoIf($[${BCHANNELINFO} = 2]?hangup:free)
exten = s,n(free),NoOp()
exten = s,n,Set(TO=${ARG1})
exten = s,n,Verbose(1,${BOUNDARY} Eingehendes Fax ${CDR(uniqueid)})
exten = s,n,Set(FAXFILE=/var/spool/fax/fax-${TO}-${CDR(uniqueid)}.tif)
exten = s,n,Set(LOCALSTATIONID=jumping frog)
exten = s,n,Answer()
exten = s,n,Wait(3)
exten = s,n,ReceiveFAX(${FAXFILE},d)
exten = s,n,Verbose(1,${BOUNDARY} Nach dem Fax!)
exten = s,n,System(/usr/local/bin/fax2mail.sh ${FAXFILE} ${TO})
;exten = s,n,capicommand(receivefax,${FAXFILE},+00497613821,Headline,k)
exten = s,n(hangup),HangUp()
exten = h,1,System(/usr/local/bin/fax2mail.sh ${FAXFILE} ${TO})
==
As you can see, the received fax file should be processed by a
bash-script, but after the call hangs up, the script is never executed.
The console log shows:
==
-- Channel 'CAPI/ISDN1#02/3821-5' FAX session '4' is complete,
result: 'SUCCESS' (FAX_SUCCESS), error: 'NO_ERROR', pages: 1,
resolution: '204x98', transfer rate: '9600', remoteSID: '4932123847885'
== Spawn extension (macro-faxin, s, 11) exited non-zero on
'CAPI/ISDN1#02/12345-5' in macro 'faxin'
== Spawn extension (capi-in, 12345, 1) exited non-zero on
'CAPI/ISDN1#02/12345-5'
== ISDN1#02: CAPI Hangingup for PLCI=0x101 in state 2
ISDN1#02: CAPI INFO 0x3490: Normal call clearing
==
Anyone seeing what I'm missing?
Thank you.
Regards,
Ruben
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Re: [asterisk-users] OT - Polycom - Which provisioning protocol to choose ?
Personally I would use HTTP too. Simple reason: You are much more flexible with it and a in most scnearios you have a webserver running anyway. I build some PHP-Script to provision SNOM VoIP phones for mass deployment and it works like a charm. Regards, Ruben -- _ -- 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] dialplan execution stops after ReceiveFax
Hi Ruben,
You should be looking at this thread
http://lists.digium.com/pipermail/asterisk-users/2011-June/263995.html
Presently I don't have the time to generate and send logs however soon
after my last post I did perform additional testing.
I am using ReceiveFAX using SPANDSP technology.
The occasions the System() call would not be executed, whether it was in
'h' of the dialplan or the main part of the macro after ReceiveFAX(),
was when a T.38 fax was being received, when it was a G.711 fax no
matter what I did to the call it would always execute the System() call
whether it was in the macro or the 'h'.
Cheers,
Larry.
Hi Larry,
Ok. It looks like I don't understand the asterisks context and macro
system deep enough.
I tried to set
==
; Faxe fuer Ruben
exten = 3821,1,Macro(faxin,ruben.roeg...@jumping-frog.org,${EXTEN})
exten = h,1,System(/usr/local/bin/fax2mail.sh ${FAXFILE} ${TO})
==
Of couse I have to set up some number filtering but now, my bash script
is beeing executed.
So, as I understand the thread you gave me to read, the h-extension is
called in the context from which the macro is called and not in the
macro itself? This could explain why it works now.
Anyway, thank you for giving me the opportunity to look beyond my own
backyard!
Regards,
Ruben
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