Re: [asterisk-users] sip_xmit: sip_xmit returned -1: Operation not permitted
Hi Jonas, I get this error when I incorrectly set my PBX gateway AND I have a sip peer trying to register outside (i.e.: a sip provider). Are you sure about your sip.conf? Giorgio Incantalupo Jonas Kellens wrote: Hello, my Asterisk CLI is flooded with the following message : [Jun 25 21:24:57] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation not permitted [Jun 25 21:25:01] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation not permitted [Jun 25 21:25:05] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation not permitted [Jun 25 21:25:09] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation not permitted [Jun 25 21:25:13] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation not permitted I have no idea where this IP comes from, there is no SIP peer or user with this IP-address. What can I do to get ride of this message that is constantly flooding my CLI ?! Reloading or restarting my Asterisk does not help ! Kind regards, 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
Re: [asterisk-users] peer IP address in CDR
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.
Retrieve such info with variables:
RTPAUDIOQOS
BRTPAUDIOQOS
And even more:
LEG1DATA
LEG2DATA
In format:
uniqueid|accountcode|chan_type|audionativeformat|audioreadformat|audiowritef
ormat|language|hangupcause|peerip|recvip|from|uri|useragent|
example:
LEG2DATA:
1277817284.0|7|SIP|alaw|alaw|alaw|en|16|192.168.0.148|192.168.0.148|sip:1003
@173test|sip:1...@192.168.0.148:5061|X-PRO build 1082
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von
Klitzing
Sent: Tuesday, June 29, 2010 6:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: 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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chan_sip.c.patch
Description: Binary data
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[asterisk-users] Echo problem in VoIP-calls
Hello list, this is the setup : analogue phone -- Grandstream GXW4008 -- Linksys WAG160N -- Asterisk-server (public) and Zoiper softphone -- Linksys WAG160N -- Asterisk-server (public) When calling with an analogue phone + Grandstream GXW and also when calling with the Zoiper softphone, we experience echo on both calling parties. Because the echo is there with the analogue phone AND with the Zoiper, I conclude that it is not the Grandstream GXW4008 gateway that is causing the echo. Could it be the router ??? These are the VoIP speed test results : VoIP test statistics Jitter: you -- server: 4.2 ms Jitter: server -- you: off Packet loss: you -- server: 0.0 % Packet loss: server -- you: off Packet discards: 0.0 % Packets out of order: 0.0 Kind regards, 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
Re: [asterisk-users] Anyone can share their config file for Cisco phone please?
Hi bruce, SIPDefault.conf #Image Version image_version:P0S3-08-8-00 #Proxy server address # Emergency Proxy info proxy_emergency: 192.168.20.4 proxy_emergency_port: 5060 # Backup Proxy info proxy_backup: 192.168.20.4 proxy_backup_port: 5060 # NAT/Firewall Traversal nat_enable: 0 nat_address: voip_control_port: 5060 start_media_port: 16384 end_media_port: 32766 nat_received_processing: 0 telnet_level: 2 # Time Server Set time zone to your location # Currently on this system the tz is GMT sntp_mode: unicast sntp_server: 192.168.20.4 time_zone: CET dst_offset: 1 dst_start_month: Mar dst_start_day: dst_start_day_of_week: Sun dst_start_week_of_month: 4 dst_start_time: 2 dst_stop_month: Oct dst_stop_day: dst_stop_day_of_week: Sun dst_stop_week_of_month: 4 dst_stop_time: 3 dst_auto_adjust: 1 enable_vad : 1 date_format : D/M/Y directory_url: http://192.168.20.4/xmlservices/phonebook.xml; logo_url: http://192.168.20.4/images/logo.bmp; SIP_MAC_ADDR.conf proxy1_address: 192.168.20.4 ; Line 1 phone number line1_name : 246 ; Line 1 name for authentication with proxy server line1_authname : 246 ; Line 1 authentication name password line1_password : afjhajshdga ; Phone Label (Text desired to be displayed in upper right corner) phone_label: XX246 i hope this help you! regards 2010/6/30 bruce bruce bruceb...@gmail.com I have an *ipphone 7965G* which has to be connected to Asterisk. It has been flashed with SIP firmware but the config file doesn't seem to work maybe I am missing something in it. I appreciate it if you can share your working sample config file with me. 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] Echo problem in VoIP-calls
Jonas Kellens wrote: Hello list, this is the setup : analogue phone -- Grandstream GXW4008 -- Linksys WAG160N -- Asterisk-server (public) and Zoiper softphone -- Linksys WAG160N -- Asterisk-server (public) When calling with an analogue phone + Grandstream GXW and also when calling with the Zoiper softphone, we experience echo on both calling parties. Because the echo is there with the analogue phone AND with the Zoiper, I conclude that it is not the Grandstream GXW4008 gateway that is causing the echo. Could it be the router ??? These are the VoIP speed test results : VoIP test statistics Jitter: you -- server: 4.2 ms Jitter: server -- you: off Packet loss: you -- server: 0.0 % Packet loss: server -- you: off Packet discards: 0.0 % Packets out of order: 0.0 Kind regards, Jonas. Echo cannot be caused by a router. The zoipher softphone is probably being used with a headset and I suspect the microphone is picking up the sounds from the earphones resulting in echo. Try turning down the earphone volume to see if this helps. If it does invest in some better headphone preferably ones where the microphone has built in background noise cancelation. For the analogue phone it could be a similar issue. Some phones are better than others. Cant you use a proper SIP phone? They work so much better. -- _ -- 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
Hello, I also thought about echo because the Zoiper softphone is used with a headset. But that didn't explain why the echo also appeared on the analogue phone + gateway. I have the same Grandstream GXW 4008 gateway with 5 analoge phones attached in another environment and there, there are no echo-problems. Can't say the analogue phones that are being used there are top of the bill, rather cheap stuff actually. When calling through the analogue phone line, there is no echo (and it seems therefore that the analogue phones that are being used meet the quality standards). The only network-element that is different in the 2 environments is the router... Jonas. On 06/30/2010 11:06 AM, Gareth Blades wrote: Echo cannot be caused by a router. The zoipher softphone is probably being used with a headset and I suspect the microphone is picking up the sounds from the earphones resulting in echo. Try turning down the earphone volume to see if this helps. If it does invest in some better headphone preferably ones where the microphone has built in background noise cancelation. For the analogue phone it could be a similar issue. Some phones are better than others. Cant you use a proper SIP phone? They work so much better. -- _ -- 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
Routers wont cause echo. In order for them to do so they would have to store the outbound voice traffic, delay it and then mix it into the inbound voice. Telephones inherently cause echo. For domestic calls the audio path is normally so short that any echo arrives back so quick the human ear does not detect it. For international calls the telco uses expensive echo cancelation technology. When you switch to voip you are often suddenly introducing a much larger delay so any excho which was present before but not noticed suddenly becomes noticable. You need to analyse the audio path your calls are taking, where the delays are being introduced and where echo cancelation is being applied. You also havent stated which end of the conversation is hearing the echo. Jonas Kellens wrote: Hello, I also thought about echo because the Zoiper softphone is used with a headset. But that didn't explain why the echo also appeared on the analogue phone + gateway. I have the same Grandstream GXW 4008 gateway with 5 analoge phones attached in another environment and there, there are no echo-problems. Can't say the analogue phones that are being used there are top of the bill, rather cheap stuff actually. When calling through the analogue phone line, there is no echo (and it seems therefore that the analogue phones that are being used meet the quality standards). The only network-element that is different in the 2 environments is the router... Jonas. On 06/30/2010 11:06 AM, Gareth Blades wrote: Echo cannot be caused by a router. The zoipher softphone is probably being used with a headset and I suspect the microphone is picking up the sounds from the earphones resulting in echo. Try turning down the earphone volume to see if this helps. If it does invest in some better headphone preferably ones where the microphone has built in background noise cancelation. For the analogue phone it could be a similar issue. Some phones are better than others. Cant you use a proper SIP phone? They work so much better. -- _ -- 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
On 30 June 2010 10:28, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, I also thought about echo because the Zoiper softphone is used with a headset. But that didn't explain why the echo also appeared on the analogue phone + gateway. It will present it self on the analogue phone when it is introduced in Zoiper. As the orignal respondent said, routers dont introduce echo. -- _ -- 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
Hello, I stated in my first post that both ends hear an echo when one speaks to the other... The only place where echo cancellation is being applied is in the Asterisk server. I have the following in sip.conf : ;-- JITTER BUFFER CONFIGURATION -- jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a ; SIP channel. Defaults to no. An enabled jitterbuffer will ; be used only if the sending side can create and the receiving ; side can not accept jitter. The SIP channel can accept jitter, ; thus a jitterbuffer on the receive SIP side will be used only ; if it is forced and enabled. jbforce = no; Forces the use of a jitterbuffer on the receive side of a SIP ; channel. Defaults to no. ;--- Thank you for your replies. Kind regards. Jonas. On 06/30/2010 11:36 AM, Gareth Blades wrote: Routers wont cause echo. In order for them to do so they would have to store the outbound voice traffic, delay it and then mix it into the inbound voice. Telephones inherently cause echo. For domestic calls the audio path is normally so short that any echo arrives back so quick the human ear does not detect it. For international calls the telco uses expensive echo cancelation technology. When you switch to voip you are often suddenly introducing a much larger delay so any excho which was present before but not noticed suddenly becomes noticable. You need to analyse the audio path your calls are taking, where the delays are being introduced and where echo cancelation is being applied. You also havent stated which end of the conversation is hearing the echo. -- _ -- 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
Thats the jitter buffer. It has no effect on echo. So you get echo when calling from the softphone to the analogue phone? What about when one of those calls somewhere else? What if they call a regular telephone number? How do you connect in order to send calls to normal phone numbers? Jonas Kellens wrote: Hello, I stated in my first post that both ends hear an echo when one speaks to the other... The only place where echo cancellation is being applied is in the Asterisk server. I have the following in sip.conf : ;-- JITTER BUFFER CONFIGURATION -- jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a ; SIP channel. Defaults to no. An enabled jitterbuffer will ; be used only if the sending side can create and the receiving ; side can not accept jitter. The SIP channel can accept jitter, ; thus a jitterbuffer on the receive SIP side will be used only ; if it is forced and enabled. jbforce = no; Forces the use of a jitterbuffer on the receive side of a SIP ; channel. Defaults to no. ;--- Thank you for your replies. Kind regards. Jonas. On 06/30/2010 11:36 AM, Gareth Blades wrote: Routers wont cause echo. In order for them to do so they would have to store the outbound voice traffic, delay it and then mix it into the inbound voice. Telephones inherently cause echo. For domestic calls the audio path is normally so short that any echo arrives back so quick the human ear does not detect it. For international calls the telco uses expensive echo cancelation technology. When you switch to voip you are often suddenly introducing a much larger delay so any excho which was present before but not noticed suddenly becomes noticable. You need to analyse the audio path your calls are taking, where the delays are being introduced and where echo cancelation is being applied. You also havent stated which end of the conversation is hearing the echo. -- _ -- 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
Hello, I did not say that the analogue phone calls the Zoiper softphone or vica versa. Calls are made to from the Zoiper to an external number like a cellphone. Calls are also made from the analogue phone to external numbers like an international number in Holland... Jonas. On 30 June 2010 10:28, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: Hello, I also thought about echo because the Zoiper softphone is used with a headset. But that didn't explain why the echo also appeared on the analogue phone + gateway. It will present it self on the analogue phone when it is introduced in Zoiper. As the orignal respondent said, routers dont introduce echo. -- _ -- 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] Adding Congestion to CDR logs
Hi, I had a breif telco outage with one of my sip providers. Is there a way to add failed calls to the cdr aswell as the connected ones? I was also thinking about having an automated process that monitored congested calls vs Succesful ones on a carrier and weight the dial plan using this. My dial plan is already run by global varialbes for day/night for landline/mobile and I was thinking that I could use the manager interface to change these variables depending on the sucess rate from an application. Not done that much research into it but I beleive that this is possible! Thanks Kenny Watson -- _ -- 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] Adding Congestion to CDR logs
Kenny Watson wrote: Hi, I had a breif telco outage with one of my sip providers. Is there a way to add failed calls to the cdr aswell as the connected ones? I was also thinking about having an automated process that monitored congested calls vs Succesful ones on a carrier and weight the dial plan using this. My dial plan is already run by global varialbes for day/night for landline/mobile and I was thinking that I could use the manager interface to change these variables depending on the sucess rate from an application. Not done that much research into it but I beleive that this is possible! Thanks Kenny Watson Yes you could certenly do that. If one of your sip providers goes down and you have qualify=yes for them then the call should fail immediatly and you can detect the return code and automatically fail over to a different provider. A better way would be to make use of AGI and write code to lookup calls costs for the specific destination so you can perform least cost routing between your providers. When the call is hungup you can record stats about that provider such as if the call failed. -- _ -- 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] RE How to break pri DID to multiple SIP Trunks
Samantha,
Are you using some type of GUI ? If you send all the traffic to a specific
context in there you can set a default route to one peer and then set
exceptions for the others. For example
[from-pri]
Exten = _X.,1,Dial(SIP/${ext...@peer1)
Exten = _X61280X,1,Dial(SIP/${ext...@peer2)
- Original Message -
From: Samantha
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, June 30, 2010 05:58
Subject: [asterisk-users] RE How to break pri DID to multiple SIP Trunks
Hey Guys
I have an indial range of 6128[01234]X being trunked sip to
xxx.yyy.189.65
Now I want to break this down into 61280x going to xxx.yyy.188.145 and
61284x going to xxx.yyy.189.199
reminder being used for fax-email etc etc etc
I have created the outbound routes and sip trunks
I can see that all the sip trunks are up
I can see the outbound routes are there and also in trunks
But it isn't working
The call gets answered by the first point xxx.yyy.189.69 and you get an rva
of the number you called is not in service
Regards
Samantha
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Re: [asterisk-users] Dial options not working
Anahi, What kind of line do you have ? POTS, PRI, SIP ? It seems like the DTMF is not coming in correctly or you have some bad settings on your end. - Original Message - From: Anahi Ludueña To: asterisk-users@lists.digium.com Sent: Wednesday, June 30, 2010 01:17 Subject: Re: [asterisk-users] Dial options not working Thanks, but I don't have any *dahdi*.conf file here... (I check in /etc/asterisk) -- Anahi Ludueña -- From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Tue, 29 Jun 2010 16:54:01 -0500 Subject: Re: [asterisk-users] Dial options not working Check your DTMF settings in *dahdi*.conf (not sure which of the dahdi files this lives in). Sounds like your DAHDI doesn’t like DTMF input. -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Tuesday, June 29, 2010 4:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dial options not working Hi, I have an extension which has the follow me option activated. The followme option should go to a IVR if no answer... The problem that I have is that everything works when I'm calling it from my extension, but if I use any landline phone or a cell phone, I'm unable to enter any options. When I press one option, it seems I do nothing... Please, could you help me? Thanks, -- Anahi Ludueña -- Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo ya! -- Dime cómo viajas y te diré qué famoso eres ¿Cuál es tu estilo, chic y deslumbrante o mundano y familiar? Descubre quién eres viajando. -- -- _ -- 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] Can't call my extension
Micholas, 1) Do you have net=yes in sip.conf ? 2) How often are you registering with the Asterisk server ? You may want to run ngrep (http://ngrep.sourceforge.net/) against the remote IP and see what happens. Chances are your router is blocking it. For ngrep you want to run something like ngrep -q -t -W byline -d any host REMOTE_IP_ADDRFESS and port 5060 Dovid - Original Message - From: Nicholas Hart To: asterisk-users@lists.digium.com Sent: Wednesday, June 30, 2010 00:23 Subject: [asterisk-users] Can't call my extension Hi, I managed to get a remote extension to work through a router which can now call all the other local extensions in asterisk. For some reason, nobody can call me back. They get failed upon trying. Keep thinking there must be some caller group to which I need be added. Or perhaps I need to add the IP address of this phone to the sip.conf file? Please let me know. Thanks. Nick -- -- _ -- 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] Adding Congestion to CDR logs
Hi Gareth, The problem I have had in the past with providers is either that the registrar is still up and its further down the line in the provider that the call is being congestied, so the qualify doesnt work! or that the providers registrar has issues but the rest of their services is up so the qualify shows the peer as down but it will still process calls (I disabled qualify for this provider). How much load would adding agi in produce, I'm processing about 2000 call attempts per hour which is going to possibly double on this box. I've been trying to keep things as light as possible. If I can get congestion into a cdr and have it sending cdr off to a SQL db it would be ideal. Thanks Kenny Support contact details: supp...@geniusgroupltd.com - Original Message - From: Gareth Blades list-aster...@skycomuk.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 30 June, 2010 11:44:58 AM Subject: Re: [asterisk-users] Adding Congestion to CDR logs Kenny Watson wrote: Hi, I had a breif telco outage with one of my sip providers. Is there a way to add failed calls to the cdr aswell as the connected ones? I was also thinking about having an automated process that monitored congested calls vs Succesful ones on a carrier and weight the dial plan using this. My dial plan is already run by global varialbes for day/night for landline/mobile and I was thinking that I could use the manager interface to change these variables depending on the sucess rate from an application. Not done that much research into it but I beleive that this is possible! Thanks Kenny Watson Yes you could certenly do that. If one of your sip providers goes down and you have qualify=yes for them then the call should fail immediatly and you can detect the return code and automatically fail over to a different provider. A better way would be to make use of AGI and write code to lookup calls costs for the specific destination so you can perform least cost routing between your providers. When the call is hungup you can record stats about that provider such as if the call failed. -- _ -- 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] Adding Congestion to CDR logs
Using standard AGI will add a fair bit of load and most of that will be due to loading the perl or php interpreter every time it is called. Your call volume is relativly high so I agree that whatever solution you go for you want to make it as streamlined as possible. Therefore I would advise that you make use of EAGI where you have a separate application process running all the time listening for connections from asterisk providing information on the calls. Since it is running all the time you dont get the startup overhead so it is purely database work. You can even have this on a separate box. You can define a short timeout so if the app does fail then your dialplan can just fail back to a safe provider. It all comes down to database queries in the end. You can query the last X number of calls to a provider made withing the last Y minutes to identify a possible problem and avoid using them and generate an alert if there is a suspected problem. Lots of things you can do once this sort of system is inplace and by having AGI set variables you can even simplify the asterisk dialplan and take load away from the asterisk box onto a separate database server which makes future expansion much easier. Kenny Watson wrote: Hi Gareth, The problem I have had in the past with providers is either that the registrar is still up and its further down the line in the provider that the call is being congestied, so the qualify doesnt work! or that the providers registrar has issues but the rest of their services is up so the qualify shows the peer as down but it will still process calls (I disabled qualify for this provider). How much load would adding agi in produce, I'm processing about 2000 call attempts per hour which is going to possibly double on this box. I've been trying to keep things as light as possible. If I can get congestion into a cdr and have it sending cdr off to a SQL db it would be ideal. Thanks Kenny Support contact details: supp...@geniusgroupltd.com - Original Message - From: Gareth Blades list-aster...@skycomuk.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 30 June, 2010 11:44:58 AM Subject: Re: [asterisk-users] Adding Congestion to CDR logs Kenny Watson wrote: Hi, I had a breif telco outage with one of my sip providers. Is there a way to add failed calls to the cdr aswell as the connected ones? I was also thinking about having an automated process that monitored congested calls vs Succesful ones on a carrier and weight the dial plan using this. My dial plan is already run by global varialbes for day/night for landline/mobile and I was thinking that I could use the manager interface to change these variables depending on the sucess rate from an application. Not done that much research into it but I beleive that this is possible! Thanks Kenny Watson Yes you could certenly do that. If one of your sip providers goes down and you have qualify=yes for them then the call should fail immediatly and you can detect the return code and automatically fail over to a different provider. A better way would be to make use of AGI and write code to lookup calls costs for the specific destination so you can perform least cost routing between your providers. When the call is hungup you can record stats about that provider such as if the call failed. -- _ -- 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] Adding Congestion to CDR logs
Hi Gareth thanks again for the responses! I defiantly think I would have to run the agi on a separate server, I'll maybe setup this in a lab. As I say the built in CDR is fine if it could include failed calls! I was planning to use a ratio of good/bad calls from a provider to determine the weighting or even the shift between good and bad! The db would be held on a separate server, all I want this box to do is handle a tonne of calls and do some transcoding on harder (I have recently bought a howler screamer card). Thanks Kenny Watson - Original Message - From: Gareth Blades list-aster...@skycomuk.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 30 June, 2010 12:14:29 PM Subject: Re: [asterisk-users] Adding Congestion to CDR logs Using standard AGI will add a fair bit of load and most of that will be due to loading the perl or php interpreter every time it is called. Your call volume is relativly high so I agree that whatever solution you go for you want to make it as streamlined as possible. Therefore I would advise that you make use of EAGI where you have a separate application process running all the time listening for connections from asterisk providing information on the calls. Since it is running all the time you dont get the startup overhead so it is purely database work. You can even have this on a separate box. You can define a short timeout so if the app does fail then your dialplan can just fail back to a safe provider. It all comes down to database queries in the end. You can query the last X number of calls to a provider made withing the last Y minutes to identify a possible problem and avoid using them and generate an alert if there is a suspected problem. Lots of things you can do once this sort of system is inplace and by having AGI set variables you can even simplify the asterisk dialplan and take load away from the asterisk box onto a separate database server which makes future expansion much easier. Kenny Watson wrote: Hi Gareth, The problem I have had in the past with providers is either that the registrar is still up and its further down the line in the provider that the call is being congestied, so the qualify doesnt work! or that the providers registrar has issues but the rest of their services is up so the qualify shows the peer as down but it will still process calls (I disabled qualify for this provider). How much load would adding agi in produce, I'm processing about 2000 call attempts per hour which is going to possibly double on this box. I've been trying to keep things as light as possible. If I can get congestion into a cdr and have it sending cdr off to a SQL db it would be ideal. Thanks Kenny Support contact details: supp...@geniusgroupltd.com - Original Message - From: Gareth Blades list-aster...@skycomuk.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 30 June, 2010 11:44:58 AM Subject: Re: [asterisk-users] Adding Congestion to CDR logs Kenny Watson wrote: Hi, I had a breif telco outage with one of my sip providers. Is there a way to add failed calls to the cdr aswell as the connected ones? I was also thinking about having an automated process that monitored congested calls vs Succesful ones on a carrier and weight the dial plan using this. My dial plan is already run by global varialbes for day/night for landline/mobile and I was thinking that I could use the manager interface to change these variables depending on the sucess rate from an application. Not done that much research into it but I beleive that this is possible! Thanks Kenny Watson Yes you could certenly do that. If one of your sip providers goes down and you have qualify=yes for them then the call should fail immediatly and you can detect the return code and automatically fail over to a different provider. A better way would be to make use of AGI and write code to lookup calls costs for the specific destination so you can perform least cost routing between your providers. When the call is hungup you can record stats about that provider such as if the call failed. -- _ -- 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
Re: [asterisk-users] What‘s the best operating syst em suggest for Asterisk 1.6.2.9
I'm not entirely sure I see where he implied it was. His answer refers to the question, I want to know what is the best OS for installing Asterisk...? I like both CentOS and Ubuntu. The next edition of the O'Reilly Asterisk book will cover installing Asterisk on both OS's. Leif. Tiago Geada wrote: Ubuntu is not Debian. I would recommend Debian tho, its rock solid and it jsut works (for anything) On 29 June 2010 12:29, Paul Belanger paul.belan...@polybeacon.com mailto:paul.belan...@polybeacon.com wrote: On Mon, Jun 28, 2010 at 10:04 PM, Zhang Shukun bit...@gmail.com mailto:bit...@gmail.com wrote: i want to know what is the best OS for install Asterisk 1.6.2.9, which should work properly on working system. Ubuntu 10.04 Server ? -- _ -- 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 options not working
Hi, do you mean what kind of extension I have? it is SIP, but from it, everything works well... In the SIP extension, the DTMF mode is rfc2833. Thanks, From: asteriskus...@dovid.net To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010 13:54:50 +0300 Subject: Re: [asterisk-users] Dial options not working Anahi, What kind of line do you have ? POTS, PRI, SIP ? It seems like the DTMF is not coming in correctly or you have some bad settings on your end. - Original Message - From: Anahi Ludueña To: asterisk-users@lists.digium.com Sent: Wednesday, June 30, 2010 01:17 Subject: Re: [asterisk-users] Dial options not working Thanks, but I don't have any *dahdi*.conf file here... (I check in /etc/asterisk) Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Tue, 29 Jun 2010 16:54:01 -0500 Subject: Re: [asterisk-users] Dial options not working Check your DTMF settings in *dahdi*.conf (not sure which of the dahdi files this lives in). Sounds like your DAHDI doesn’t like DTMF input. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Tuesday, June 29, 2010 4:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dial options not working Hi, I have an extension which has the follow me option activated. The followme option should go to a IVR if no answer... The problem that I have is that everything works when I'm calling it from my extension, but if I use any landline phone or a cell phone, I'm unable to enter any options. When I press one option, it seems I do nothing... Please, could you help me? Thanks, Anahi Ludueña Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo ya! Dime cómo viajas y te diré qué famoso eres ¿Cuál es tu estilo, chic y deslumbrante o mundano y familiar? Descubre quién eres viajando. -- _ -- 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 _ ¿Quieres descubrir todos los trucos de Windows 7? ¡Hazlo aquí! http://www.sietesunpueblodeexpertos.com/index_windows7.html-- _ -- 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 options not working
Hi, Have you tried sending the dtmf inband? I've had more success interoping betwen different vendors with inband DTMF. Thanks Kenny Watson Kenny Watson From: Anahi Ludueña a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Sent: Wednesday, 30 June, 2010 12:50:23 PM Subject: Re: [asterisk-users] Dial options not working Hi, do you mean what kind of extension I have? it is SIP, but from it, everything works well... In the SIP extension, the DTMF mode is rfc2833. Thanks, From: asteriskus...@dovid.net To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010 13:54:50 +0300 Subject: Re: [asterisk-users] Dial options not working Anahi, What kind of line do you have ? POTS, PRI, SIP ? It seems like the DTMF is not coming in correctly or you have some bad settings on your end. - Original Message - From: Anahi Ludueña To: asterisk-users@lists.digium.com Sent: Wednesday, June 30, 2010 01:17 Subject: Re: [asterisk-users] Dial options not working Thanks, but I don't have any * dahdi *.conf file here... (I check in /etc/asterisk) Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Tue, 29 Jun 2010 16:54:01 -0500 Subject: Re: [asterisk-users] Dial options not working Check your DTMF settings in * dahdi *.conf (not sure which of the dahdi files this lives in). Sounds like your DAHDI doesn’t like DTMF input. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Tuesday, June 29, 2010 4:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dial options not working Hi, I have an extension which has the follow me option activated. The followme option should go to a IVR if no answer... The problem that I have is that everything works when I'm calling it from my extension, but if I use any landline phone or a cell phone, I'm unable to enter any options. When I press one option, it seems I do nothing... Please, could you help me? Thanks, Anahi Ludueña Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo ya! Dime cómo viajas y te diré qué famoso eres ¿Cuál es tu estilo, chic y deslumbrante o mundano y familiar? Descubre quién eres viajando. -- _ -- 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 Del Lado Oscuro de Internet protegerte puedes. ¡Entra ya en www.ayudartepodria.com! -- _ -- 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] Dial options not working
Hi, yes, I've just tried to use the dtmf mode inband, but it doesn't work with landline phones or cell phones... Thanks, Anahi Ludueña Date: Wed, 30 Jun 2010 12:56:59 +0100 From: kwat...@geniusgroupltd.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dial options not working Hi, Have you tried sending the dtmf inband? I've had more success interoping betwen different vendors with inband DTMF. Thanks Kenny Watson From: Anahi Ludueña a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Sent: Wednesday, 30 June, 2010 12:50:23 PM Subject: Re: [asterisk-users] Dial options not working Hi, do you mean what kind of extension I have? it is SIP, but from it, everything works well... In the SIP extension, the DTMF mode is rfc2833. Thanks, From: asteriskus...@dovid.net To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010 13:54:50 +0300 Subject: Re: [asterisk-users] Dial options not working Anahi, What kind of line do you have ? POTS, PRI, SIP ? It seems like the DTMF is not coming in correctly or you have some bad settings on your end. - Original Message - From: Anahi Ludueña To: asterisk-users@lists.digium.com Sent: Wednesday, June 30, 2010 01:17 Subject: Re: [asterisk-users] Dial options not working Thanks, but I don't have any *dahdi*.conf file here... (I check in /etc/asterisk) Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Tue, 29 Jun 2010 16:54:01 -0500 Subject: Re: [asterisk-users] Dial options not working Check your DTMF settings in *dahdi*.conf (not sure which of the dahdi files this lives in). Sounds like your DAHDI doesn’t like DTMF input. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Tuesday, June 29, 2010 4:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dial options not working Hi, I have an extension which has the follow me option activated. The followme option should go to a IVR if no answer... The problem that I have is that everything works when I'm calling it from my extension, but if I use any landline phone or a cell phone, I'm unable to enter any options. When I press one option, it seems I do nothing... Please, could you help me? Thanks, Anahi Ludueña Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo ya! Dime cómo viajas y te diré qué famoso eres ¿Cuál es tu estilo, chic y deslumbrante o mundano y familiar? Descubre quién eres viajando. -- _ -- 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 Del Lado Oscuro de Internet protegerte puedes. ¡Entra ya en www.ayudartepodria.com! -- _ -- 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 _ Sé el protagonista de GQ con Messenger y Vodafone Blackberry. ¡Y gana premios! http://serviciosmoviles.es.msn.com/messenger/vodafone.aspx-- _ -- 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] 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
[asterisk-users] Minimum modules required to run VoIP and CDR
What is the minimal module set required to run SIP with database CDR logging. I compiled Asterisk from source and I obviously compiled more stuff than I needed for VoIP and CDR logging to postgres. Sometimes there is a long gap between Asterisk starting and devices being able to register. sip commands do not work, and they appear to be the last items to be loaded. Is there some way of checking the module dependencies and removing those not needed? Can the modules be interrogated to find out their dependencies, probably starting with chan_sip and some cdr and database modules? -- _ -- 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 Delay with remote stations?
this can be cause if you are using an ADSL link with your remote phones .. or maybe some 3G networks can cause that delay in the first response as the ACK message will be late to arrive and if the delay was too high .. the call will drop.one more thing if your remote phones are (Queue Members) this can be caused by a configuration of the queue itself something related to memberdelay directive. try setting it to 0 or something similar.Regards -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 386 492 9993 From: william.stillwell-li...@ablebody.net To: asterisk-users@lists.digium.com Date: Tue, 29 Jun 2010 10:06:55 -0400 Subject: [asterisk-users] SIP Delay with remote stations? I have several remote phones that experience a slight “call” delay when answering phones, ie, they will answer, speak a few words, and then the remote caller will hear them, and the first half is cutoff? Any idea what could be causing this? Thanks, Bill. _ The New Busy think 9 to 5 is a cute idea. Combine multiple calendars with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multicalendarocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_5-- _ -- 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
On 06/30/2010 12:20 PM, Gareth Blades wrote: So you get echo when calling from the softphone to the analogue phone? From softphone to analogue phone is echo. What if they call a regular telephone number? Calling to a cellphone number or a fixed number on another Telco-network : echo How do you connect in order to send calls to normal phone numbers? The network setup is : analogue+GXW / softphone -- Linksys WAG160N -- Asterisk server -- ITSP -- other networks So basically, there's always an echo. 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
Re: [asterisk-users] Echo problem in VoIP-calls
Jonas Kellens wrote: On 06/30/2010 12:20 PM, Gareth Blades wrote: So you get echo when calling from the softphone to the analogue phone? From softphone to analogue phone is echo. What if they call a regular telephone number? Calling to a cellphone number or a fixed number on another Telco-network : echo How do you connect in order to send calls to normal phone numbers? The network setup is : analogue+GXW / softphone -- Linksys WAG160N -- Asterisk server -- ITSP -- other networks So basically, there's always an echo. Jonas. By ITSP do you mean a SIP provider? -- _ -- 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
Internet Telephony Service Provider = SIP provider. The company that connects the Asterisk-server via a SIP trunk with the other networks like GSM, analogue carriers... Jonas. By ITSP do you mean a SIP provider? -- _ -- 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
On 30 Jun 2010, at 13:48, Gareth Blades wrote: By ITSP do you mean a SIP provider? ITSP: Internet Telephony Service Provider S -- _ -- 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] Anyone can share their config file for Cisco phone please?
Thanks a lot. -Bruce On Wed, Jun 30, 2010 at 4:55 AM, Emanuele Carbone carbe...@gmail.comwrote: Hi bruce, SIPDefault.conf #Image Version image_version:P0S3-08-8-00 #Proxy server address # Emergency Proxy info proxy_emergency: 192.168.20.4 proxy_emergency_port: 5060 # Backup Proxy info proxy_backup: 192.168.20.4 proxy_backup_port: 5060 # NAT/Firewall Traversal nat_enable: 0 nat_address: voip_control_port: 5060 start_media_port: 16384 end_media_port: 32766 nat_received_processing: 0 telnet_level: 2 # Time Server Set time zone to your location # Currently on this system the tz is GMT sntp_mode: unicast sntp_server: 192.168.20.4 time_zone: CET dst_offset: 1 dst_start_month: Mar dst_start_day: dst_start_day_of_week: Sun dst_start_week_of_month: 4 dst_start_time: 2 dst_stop_month: Oct dst_stop_day: dst_stop_day_of_week: Sun dst_stop_week_of_month: 4 dst_stop_time: 3 dst_auto_adjust: 1 enable_vad : 1 date_format : D/M/Y directory_url: http://192.168.20.4/xmlservices/phonebook.xml; logo_url: http://192.168.20.4/images/logo.bmp; SIP_MAC_ADDR.conf proxy1_address: 192.168.20.4 ; Line 1 phone number line1_name : 246 ; Line 1 name for authentication with proxy server line1_authname : 246 ; Line 1 authentication name password line1_password : afjhajshdga ; Phone Label (Text desired to be displayed in upper right corner) phone_label: XX246 i hope this help you! regards 2010/6/30 bruce bruce bruceb...@gmail.com I have an *ipphone 7965G* which has to be connected to Asterisk. It has been flashed with SIP firmware but the config file doesn't seem to work maybe I am missing something in it. I appreciate it if you can share your working sample config file with me. 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 -- _ -- 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
The DNS setup itself is fine. The sip module just seems to take too much time to load. My modules.conf uses autoload=yes and it seems that many unwanted modules are loaded before sip itself starts. On 30 June 2010 13:52, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: 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 -- _ -- 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
Jonas Kellens wrote: Internet Telephony Service Provider = SIP provider. The company that connects the Asterisk-server via a SIP trunk with the other networks like GSM, analogue carriers... Jonas. By ITSP do you mean a SIP provider? Thats where I believe the problem lies. You are sending audio to them and they are putting it onto the PSTN network. When the audio comes back from the PSTN it has echo on it. They are not performing echo cancellation. If it is an international call from the ITSP's perspective then teh network operator should be performing echo cancelation anyway. If its a national call then the telco doesnt perform echo cancelation but the ITSP should do it themselves. The only time this is not needed is if the phones have a very low delay to the ITSP but since this is normally not the case echo cancelation must be performed at this point. -- _ -- 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
Gareth, multiple users/SIP-accounts use this asterisk server from many locations. Like I said: in another location with a similar setup, there are no echo-complaints on received or made calls. If you say that it has nothing to do with the Cisco-router, I don't really know what to go looking for... I will take your advise and try with a SIP-phone (snom 320). What do I do if : 1. I also have echo with a SIP-phone ? 2. I do not have echo with a SIP-phone ? Jonas. On 06/30/2010 03:52 PM, Gareth Blades wrote: Thats where I believe the problem lies. You are sending audio to them and they are putting it onto the PSTN network. When the audio comes back from the PSTN it has echo on it. They are not performing echo cancellation. If it is an international call from the ITSP's perspective then teh network operator should be performing echo cancelation anyway. If its a national call then the telco doesnt perform echo cancelation but the ITSP should do it themselves. The only time this is not needed is if the phones have a very low delay to the ITSP but since this is normally not the case echo cancelation must be performed at this point. -- _ -- 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] queue command in asterisk 1.4 with macro-argument
Hello list, I notice on the wiki that it is possible to execute a macro or a gosub within the queue-command in asterisk 1.6.x 1. Does this mean the macro/gosub is executed everytime a queued call is answered by a queue member ? 2. I'm using asterisk 1.4.30. Is there a backport or other way to make use of this 1.6-functionality ?? Kind regards, 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
Re: [asterisk-users] Echo problem in VoIP-calls
Try the SIP phone. If it is better then you might try looking to see if there are any echo cancelation settings on the softphone or analogue adapter you can change. Try turning echo cancelation off aswell since if there are two running they can interfere with each other and make the situation worse. If you hear echo on that phone then it might be that the network connection from that location has a higher latency making the echo far more noticeable. If the other party you are connecting to hears echo then this could be down to the phone or the jitter buffer. If you start with a small jitter buffer the echo cancelation will train to that but if you get increased jitter the buffer will grow and add an additional delay to the audio. Often echo cancelation only trains at the start of a call. Maybe try disabling the jitter buffer. Jonas Kellens wrote: Gareth, multiple users/SIP-accounts use this asterisk server from many locations. Like I said: in another location with a similar setup, there are no echo-complaints on received or made calls. If you say that it has nothing to do with the Cisco-router, I don't really know what to go looking for... I will take your advise and try with a SIP-phone (snom 320). What do I do if : 1. I also have echo with a SIP-phone ? 2. I do not have echo with a SIP-phone ? Jonas. On 06/30/2010 03:52 PM, Gareth Blades wrote: Thats where I believe the problem lies. You are sending audio to them and they are putting it onto the PSTN network. When the audio comes back from the PSTN it has echo on it. They are not performing echo cancellation. If it is an international call from the ITSP's perspective then teh network operator should be performing echo cancelation anyway. If its a national call then the telco doesnt perform echo cancelation but the ITSP should do it themselves. The only time this is not needed is if the phones have a very low delay to the ITSP but since this is normally not the case echo cancelation must be performed at this point. -- _ -- 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] How to work Asterisk with Video Conference
Hi, I have installed Asterisk 1.6. I have to configure Asterisk as a Video Conferancing purpose. What package I need to configure and what steps I need to follow to configure in dialplan to authenticate user. Regards, Hiren Mistry -- _ -- 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
Will turning off the jitter buffer affect the quality of the other calls ?? jbenable = no I must say I'm not really into these jitter-settings in asterisk. I made jbenable=yes as it can do no harm... Jonas. On 06/30/2010 04:24 PM, Gareth Blades wrote: Try the SIP phone. If it is better then you might try looking to see if there are any echo cancelation settings on the softphone or analogue adapter you can change. Try turning echo cancelation off aswell since if there are two running they can interfere with each other and make the situation worse. If you hear echo on that phone then it might be that the network connection from that location has a higher latency making the echo far more noticeable. If the other party you are connecting to hears echo then this could be down to the phone or the jitter buffer. If you start with a small jitter buffer the echo cancelation will train to that but if you get increased jitter the buffer will grow and add an additional delay to the audio. Often echo cancelation only trains at the start of a call. Maybe try disabling the jitter buffer. -- _ -- 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] queue command in asterisk 1.4 with macro-argument
Yes it gets called when the call is connected to a queue member. In version 1.4.x you can execute an AGI instead of a sub or macro. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 30, 2010, at 7:20 AM, Jonas Kellens wrote: Hello list, I notice on the wiki that it is possible to execute a macro or a gosub within the queue-command in asterisk 1.6.x 1. Does this mean the macro/gosub is executed everytime a queued call is answered by a queue member ? 2. I'm using asterisk 1.4.30. Is there a backport or other way to make use of this 1.6-functionality ?? Kind regards, 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
Re: [asterisk-users] Echo problem in VoIP-calls
The harm in any of these settings is environmentally controlled. What does no harm in one setup can be a deal breaker on a smaller machine or slightly different technology. How harmful or harmless jbenable is depends on your hardware and what your other settings are. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, June 30, 2010 9:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Echo problem in VoIP-calls Will turning off the jitter buffer affect the quality of the other calls ?? jbenable = no I must say I'm not really into these jitter-settings in asterisk. I made jbenable=yes as it can do no harm... Jonas. On 06/30/2010 04:24 PM, Gareth Blades wrote: Try the SIP phone. If it is better then you might try looking to see if there are any echo cancelation settings on the softphone or analogue adapter you can change. Try turning echo cancelation off aswell since if there are two running they can interfere with each other and make the situation worse. If you hear echo on that phone then it might be that the network connection from that location has a higher latency making the echo far more noticeable. If the other party you are connecting to hears echo then this could be down to the phone or the jitter buffer. If you start with a small jitter buffer the echo cancelation will train to that but if you get increased jitter the buffer will grow and add an additional delay to the audio. Often echo cancelation only trains at the start of a call. Maybe try disabling the jitter buffer. -- _ -- 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] queue command in asterisk 1.4 withmacro-argument
This gives you some flexibility and change-proofing that a back-port will not. Since gosub is a depreciation candidate, you can use the AGI to either run the macro or do the macro functionality internally. I'm a HUGE fan of AGI, but keeping things in the dialplan is a better option when you can. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson Sent: Wednesday, June 30, 2010 9:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] queue command in asterisk 1.4 withmacro-argument Yes it gets called when the call is connected to a queue member. In version 1.4.x you can execute an AGI instead of a sub or macro. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 30, 2010, at 7:20 AM, Jonas Kellens wrote: Hello list, I notice on the wiki that it is possible to execute a macro or a gosub within the queue-command in asterisk 1.6.x 1. Does this mean the macro/gosub is executed everytime a queued call is answered by a queue member ? 2. I'm using asterisk 1.4.30. Is there a backport or other way to make use of this 1.6-functionality ?? Kind regards, 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
Re: [asterisk-users] Echo problem in VoIP-calls
Yes if you have a link where there is a lot of jitter it may affect the call quality. I would try turning it off to see if it cures the problem and if it does then you can restore the setting and implement a workaround. Jonas Kellens wrote: Will turning off the jitter buffer affect the quality of the other calls ?? jbenable = no I must say I'm not really into these jitter-settings in asterisk. I made jbenable=yes as it can do no harm... Jonas. On 06/30/2010 04:24 PM, Gareth Blades wrote: Try the SIP phone. If it is better then you might try looking to see if there are any echo cancelation settings on the softphone or analogue adapter you can change. Try turning echo cancelation off aswell since if there are two running they can interfere with each other and make the situation worse. If you hear echo on that phone then it might be that the network connection from that location has a higher latency making the echo far more noticeable. If the other party you are connecting to hears echo then this could be down to the phone or the jitter buffer. If you start with a small jitter buffer the echo cancelation will train to that but if you get increased jitter the buffer will grow and add an additional delay to the audio. Often echo cancelation only trains at the start of a call. Maybe try disabling the jitter buffer. -- _ -- 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] Anyone can share their config file for Cisco phone please?
On Wed, Jun 30, 2010 at 8:40 AM, bruce bruce bruceb...@gmail.com wrote: Thanks a lot. -Bruce On Wed, Jun 30, 2010 at 4:55 AM, Emanuele Carbone carbe...@gmail.comwrote: Hi bruce, SIPDefault.conf I think you need one of the newer XML config files for the 7965. I have an example that works with a 7941 on my website (you can find the link my signature), I think with a little adaptation you can make it work with a 7965. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- 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] Detecting hook flash in asterisk
Hi Paul, On Sat, Jun 26, 2010 at 1:33 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sat, Jun 26, 2010 at 7:33 AM, Deepesh D deep.d2...@gmail.com wrote: Is it possible to do this action on hook flash? Currently no. You would need to add logic to the channel driver. Or use DTMF to initiate the hookflash: My PSTN line has call waiting, and I have to use zapflash application to answer the new incoming call. If I want to flash hook to switch calls, which channel driver do I need to look at? chan_dahdi? I noticed that I can use hook flash to switch between SIP calls, or even between a SIP call and a PSTN call, does this mean chan_sip has such hook flash detection logic so I can learn from there? extensions.conf [globals] DYNAMIC_FEATURES=zapflash features.conf [applicationmap] zapflash = *0,callee,flash,() -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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 -- Ye Liu (AKA @jaux) http://jaux.net -- _ -- 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 1.6 and multiple parking
Hi, Thanks, I thought I could find out about that without installing 1.6, but in the end I did install it on a test server and it answered a few questions. One thing though: I can park calls, in separate private lots, but I can never pick them up again. I have context = some_context defined in features.conf (under the private parking lot) and an include = some_context at the right place, and when I park a call exten = 800 (thats what I use) appears correctly. But I can't seem to pick it up, when I dial 800 it says I am sorry there is no call parked on that extension. This is the relevant context when a call is parked. It clearly shows a call being parked. localhost*CLI dialplan show parkingtest [ Context 'parkingtest' created by 'features' ] '700' = 1. Park() [features] '800' = 1. ParkedCall(800)[features] And the features.conf snippet (everything else is default features.conf from 1.6): [parkinglot_test] context = parkingtest parkpos = 800-805 findslot = next What am I missing you think? I only set the CHANNEL(parkinglot) value when parking the call. Do I need to set that value when picking up a call? (after all, I have no accessz to extension 800 it is created by features.conf) Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aksel Celasun Sent: Monday, June 28, 2010 16:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking Hello there You should have a look at features.conf Regards Aksel Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike Sendt: 28. juni 2010 21:39 Til: 'Asterisk Users Mailing List - Non-Commercial Discussion' Emne: [asterisk-users] Asterisk 1.6 and multiple parking Hi, One of the big features of 1.6 was described as multi-tenant parking. Basically, parking people in different lots so the sales dept. could only pick up their calls, and tech support theirs and no mix up was possible. I can only find the original announcement and others asking the same question. Is there some sort of sample conf file of how I would get this functionnal on the latest 1.6.x? Regards, Mike -- _ -- 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
On Wed, Jun 30, 2010 at 8:50 AM, Frank Church voi...@googlemail.com wrote: The DNS setup itself is fine. The sip module just seems to take too much time to load. My modules.conf uses autoload=yes and it seems that many unwanted modules are loaded before sip itself starts. You can stop asterisk and then start it again with asterisk -cvv to see a list of everything that starts up. Take that list, tweak it, then reconfigure your modules.conf to only load the ones you want. Then start it again using the same procedure, making note of any errors that pop up, and resolve them. It will take a little trial and error, but you should be able to get it done. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- 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] Problem in establish call from a2billing users.
Hi All, I installed a2billing with asterisk FreePBX . I can able to login and make a call with FreePBX but when i am using the users which is created in a2billing the call was not established . I know somewhere i missed the configuration please any one help me to resolve this issue . Thanks in advance. regards, gokul., -- _ -- 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 1.6 and multiple parking
Actually, I should simply have tried. I did need to set CHANNEL(parkinglot). I may have some more questions, but at least it's working right now, and use my own custom extension to pickup the calls. So basically I don't need to (or even can!) include the parking context, I need to setup the extensions myself. For futur reference. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, June 30, 2010 11:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking Hi, Thanks, I thought I could find out about that without installing 1.6, but in the end I did install it on a test server and it answered a few questions. One thing though: I can park calls, in separate private lots, but I can never pick them up again. I have context = some_context defined in features.conf (under the private parking lot) and an include = some_context at the right place, and when I park a call exten = 800 (thats what I use) appears correctly. But I can't seem to pick it up, when I dial 800 it says I am sorry there is no call parked on that extension. This is the relevant context when a call is parked. It clearly shows a call being parked. localhost*CLI dialplan show parkingtest [ Context 'parkingtest' created by 'features' ] '700' = 1. Park() [features] '800' = 1. ParkedCall(800)[features] And the features.conf snippet (everything else is default features.conf from 1.6): [parkinglot_test] context = parkingtest parkpos = 800-805 findslot = next What am I missing you think? I only set the CHANNEL(parkinglot) value when parking the call. Do I need to set that value when picking up a call? (after all, I have no accessz to extension 800 it is created by features.conf) Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aksel Celasun Sent: Monday, June 28, 2010 16:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking Hello there You should have a look at features.conf Regards Aksel Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike Sendt: 28. juni 2010 21:39 Til: 'Asterisk Users Mailing List - Non-Commercial Discussion' Emne: [asterisk-users] Asterisk 1.6 and multiple parking Hi, One of the big features of 1.6 was described as multi-tenant parking. Basically, parking people in different lots so the sales dept. could only pick up their calls, and tech support theirs and no mix up was possible. I can only find the original announcement and others asking the same question. Is there some sort of sample conf file of how I would get this functionnal on the latest 1.6.x? Regards, Mike -- _ -- 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] Problem with extensions in IVR and queues
Hi people, we have some extensions which are included in the IVRs and/or queues. Everything works fine, but the calls done from these extensions are hang up after 30 o 35 seconds. If they are not included in the IVR or queues, the calls are performed well. Do you know if there is something else to set? Thanks, Anahi Ludueña _ ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! www.ayudartepodria.com-- _ -- 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] queue command in asterisk 1.4 with macro-argument
Taking my first steps into AGI then :
[r...@asterisk agi-bin]# cat sample.agi
#!/usr/bin/php -q
?php
$MYSQLSERVER2=localhost;
$MYSQLUSER2=user;
$MYSQLPASSWD2=passwd;
set_time_limit(30);
require('phpagi/phpagi.php');
$agi = new AGI();
$db=mysql_connect($MYSQLSERVER2, $MYSQLUSER2, $MYSQLPASSWD2);
mysql_select_db(Asterisk, $db);
$QUERY=SELECT vmcontext FROM AstDB WHERE ID='40';
$agi-verbose(query is: $QUERY, 3);
$result=mysql_query($QUERY);
$VMCONTEXT=mysql_fetch_array($result);
$agi-verbose(VMCONTEXT is: $VMCONTEXT, 3);
$vmcontext=$VMCONTEXT['vmcontext'];
$exten = $agi-request['agi_extension']; //Dialed extension
// the result is stored in $exten
$agi-verbose(variable exten : $exten, 3);
$agi-verbose(variable vmcontext : $vmcontext, 3);
//
?
[Jun 30 17:26:04] -- Executing [...@test:3] AGI(SIP/test-0054,
sample.agi) in new stack
[Jun 30 17:26:04] -- Launched AGI Script
/var/lib/asterisk/agi-bin/sample.agi
[Jun 30 17:26:04] -- sample.agi: query is: SELECT vmcontext FROM
AstDB WHERE klantID='40'
[Jun 30 17:26:04] -- sample.agi: VMCONTEXT is:
[Jun 30 17:26:04] -- sample.agi: variable exten : 123
[Jun 30 17:26:04] -- sample.agi: variable vmcontext :
[Jun 30 17:26:04] -- AGI Script sample.agi completed, returning 0
Does AGI not interpret my query correctly ? As there is no output for
$vmcontext...
Jonas.
On 06/30/2010 04:54 PM, Jim Dickenson wrote:
Yes it gets called when the call is connected to a queue member.
In version 1.4.x you can execute an AGI instead of a sub or macro.
--
_
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Re: [asterisk-users] Problem with extensions in IVR and queues
Sounds like you are getting a dial without bridge asterisk dials x and make the connection, but because the bridge doesnt happen for what ever reason, the call disconnects like no one ever answered. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Wednesday, June 30, 2010 10:29 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problem with extensions in IVR and queues Hi people, we have some extensions which are included in the IVRs and/or queues. Everything works fine, but the calls done from these extensions are hang up after 30 o 35 seconds. If they are not included in the IVR or queues, the calls are performed well. Do you know if there is something else to set? Thanks, _ Anahi Ludueña _ ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! -- _ -- 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] queue command in asterisk 1.4with macro-argument
1. (personal preference) I wouldn't use PHP
2. that out of the way, I comment out the AGI stuff and run my AGI's
from bash to make sure the non AGI stuff is happy.
3. the AGI seems to be ok here, I'd make sure my SQL stuff is good.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, June 30, 2010 10:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] queue command in asterisk 1.4with
macro-argument
Taking my first steps into AGI then :
[r...@asterisk agi-bin]# cat sample.agi
#!/usr/bin/php -q
?php
$MYSQLSERVER2=localhost;
$MYSQLUSER2=user;
$MYSQLPASSWD2=passwd;
set_time_limit(30);
require('phpagi/phpagi.php');
$agi = new AGI();
$db=mysql_connect($MYSQLSERVER2, $MYSQLUSER2, $MYSQLPASSWD2);
mysql_select_db(Asterisk, $db);
$QUERY=SELECT vmcontext FROM AstDB WHERE ID='40';
$agi-verbose(query is: $QUERY, 3);
$result=mysql_query($QUERY);
$VMCONTEXT=mysql_fetch_array($result);
$agi-verbose(VMCONTEXT is: $VMCONTEXT, 3);
$vmcontext=$VMCONTEXT['vmcontext'];
$exten = $agi-request['agi_extension']; //Dialed extension
// the result is stored in $exten
$agi-verbose(variable exten : $exten, 3);
$agi-verbose(variable vmcontext : $vmcontext, 3);
//
?
[Jun 30 17:26:04] -- Executing [...@test:3] AGI(SIP/test-0054,
sample.agi) in new stack
[Jun 30 17:26:04] -- Launched AGI Script
/var/lib/asterisk/agi-bin/sample.agi
[Jun 30 17:26:04] -- sample.agi: query is: SELECT vmcontext FROM AstDB
WHERE klantID='40'
[Jun 30 17:26:04] -- sample.agi: VMCONTEXT is:
[Jun 30 17:26:04] -- sample.agi: variable exten : 123
[Jun 30 17:26:04] -- sample.agi: variable vmcontext :
[Jun 30 17:26:04] -- AGI Script sample.agi completed, returning 0
Does AGI not interpret my query correctly ? As there is no output for
$vmcontext...
Jonas.
On 06/30/2010 04:54 PM, Jim Dickenson wrote:
Yes it gets called when the call is connected to a queue member.
In version 1.4.x you can execute an AGI instead of a sub or macro.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
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Re: [asterisk-users] queue command in asterisk 1.4 with macro-argument
Here is a simple AGI using cagi that creates a user event when a call is
connected with a queue member:
#include stdio.h
#include stdarg.h
#include cagi.h
int main (int argc, char *argv[]) {
AGI_TOOLS agi;
AGI_CMD_RESULT res;
intrtn;
char channel_name[200], uniqueid[200], Interface[200], Event[1000];
rtn = AGITool_Init(agi);
// rtn = AGITool_verbose(agi, res, AGITool_ListGetVal(agi.agi_vars,
// agi_request), 0);
// sprintf(Event, Do verbose= %d, rtn);
// AGITool_verbose(agi, res, Event, 0);
rtn = AGITool_get_variable2(agi, res, CHANNEL,
channel_name, sizeof(channel_name));
// sprintf(Event, Get CHANNEL = %d, rtn);
// AGITool_verbose(agi, res, Event, 0);
rtn = AGITool_get_variable2(agi, res, UNIQUEID,
uniqueid, sizeof(uniqueid));
// sprintf(Event, Get UNIQUEID = %d, rtn);
// AGITool_verbose(agi, res, Event, 0);
rtn = AGITool_get_variable2(agi, res, MEMBERINTERFACE,
Interface, sizeof(Interface));
// sprintf(Event, Get MEMBERINTERFACE = %d, rtn);
// AGITool_verbose(agi, res, Event, 0);
sprintf(Event, DidQueue|\%s %s %s, uniqueid, channel_name,
Interface);
rtn = AGITool_exec(agi, res, UserEvent, Event);
// sprintf(Event, Do UserEvent = %d, rtn);
// AGITool_verbose(agi, res, Event, 0);
AGITool_Destroy(agi);
return 0;
} /* main */
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jun 30, 2010, at 8:31 AM, Jonas Kellens wrote:
Taking my first steps into AGI then :
[r...@asterisk agi-bin]# cat sample.agi
#!/usr/bin/php -q
?php
$MYSQLSERVER2=localhost;
$MYSQLUSER2=user;
$MYSQLPASSWD2=passwd;
set_time_limit(30);
require('phpagi/phpagi.php');
$agi = new AGI();
$db=mysql_connect($MYSQLSERVER2, $MYSQLUSER2, $MYSQLPASSWD2);
mysql_select_db(Asterisk, $db);
$QUERY=SELECT vmcontext FROM AstDB WHERE ID='40';
$agi-verbose(query is: $QUERY, 3);
$result=mysql_query($QUERY);
$VMCONTEXT=mysql_fetch_array($result);
$agi-verbose(VMCONTEXT is: $VMCONTEXT, 3);
$vmcontext=$VMCONTEXT['vmcontext'];
$exten = $agi-request['agi_extension']; //Dialed extension
// the result is stored in $exten
$agi-verbose(variable exten : $exten, 3);
$agi-verbose(variable vmcontext : $vmcontext, 3);
//
?
[Jun 30 17:26:04] -- Executing [...@test:3] AGI(SIP/test-0054,
sample.agi) in new stack
[Jun 30 17:26:04] -- Launched AGI Script
/var/lib/asterisk/agi-bin/sample.agi
[Jun 30 17:26:04] -- sample.agi: query is: SELECT vmcontext FROM AstDB
WHERE klantID='40'
[Jun 30 17:26:04] -- sample.agi: VMCONTEXT is:
[Jun 30 17:26:04] -- sample.agi: variable exten : 123
[Jun 30 17:26:04] -- sample.agi: variable vmcontext :
[Jun 30 17:26:04] -- AGI Script sample.agi completed, returning 0
Does AGI not interpret my query correctly ? As there is no output for
$vmcontext...
Jonas.
On 06/30/2010 04:54 PM, Jim Dickenson wrote:
Yes it gets called when the call is connected to a queue member.
In version 1.4.x you can execute an AGI instead of a sub or macro.
--
_
-- 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 1.6 and multiple parking
Here is my only question left about parkinglots in 1.6. How does the parkinghints=yes parameter work? I've tried using core show hints , but there are never any hints. Even when a call is actually parked in the correct parking lot. Any tips? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, June 30, 2010 11:27 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking Actually, I should simply have tried. I did need to set CHANNEL(parkinglot). I may have some more questions, but at least it's working right now, and use my own custom extension to pickup the calls. So basically I don't need to (or even can!) include the parking context, I need to setup the extensions myself. For futur reference. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, June 30, 2010 11:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking Hi, Thanks, I thought I could find out about that without installing 1.6, but in the end I did install it on a test server and it answered a few questions. One thing though: I can park calls, in separate private lots, but I can never pick them up again. I have context = some_context defined in features.conf (under the private parking lot) and an include = some_context at the right place, and when I park a call exten = 800 (thats what I use) appears correctly. But I can't seem to pick it up, when I dial 800 it says I am sorry there is no call parked on that extension. This is the relevant context when a call is parked. It clearly shows a call being parked. localhost*CLI dialplan show parkingtest [ Context 'parkingtest' created by 'features' ] '700' = 1. Park() [features] '800' = 1. ParkedCall(800)[features] And the features.conf snippet (everything else is default features.conf from 1.6): [parkinglot_test] context = parkingtest parkpos = 800-805 findslot = next What am I missing you think? I only set the CHANNEL(parkinglot) value when parking the call. Do I need to set that value when picking up a call? (after all, I have no accessz to extension 800 it is created by features.conf) Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aksel Celasun Sent: Monday, June 28, 2010 16:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking Hello there You should have a look at features.conf Regards Aksel Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike Sendt: 28. juni 2010 21:39 Til: 'Asterisk Users Mailing List - Non-Commercial Discussion' Emne: [asterisk-users] Asterisk 1.6 and multiple parking Hi, One of the big features of 1.6 was described as multi-tenant parking. Basically, parking people in different lots so the sales dept. could only pick up their calls, and tech support theirs and no mix up was possible. I can only find the original announcement and others asking the same question. Is there some sort of sample conf file of how I would get this functionnal on the latest 1.6.x? Regards, Mike -- _ -- 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] queue command in asterisk 1.4with macro-argument
Danny, 1. I only know php, I'm no programmer 3. the query works in normal PHP. Can I debug to know what's going wrong ? Jonas. On 06/30/2010 05:42 PM, Danny Nicholas wrote: 1. (personal preference) I wouldn't use PHP 2. that out of the way, I comment out the AGI stuff and run my AGI's from bash to make sure the non AGI stuff is happy. 3. the AGI seems to be ok here, I'd make sure my SQL stuff is good. -- _ -- 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 1.6 and multiple parking
In 1.4 you set up the lots you want to monitor as hints; not sure how this works in 1.6. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, June 30, 2010 11:24 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking Here is my only question left about parkinglots in 1.6. How does the parkinghints=yes parameter work? I've tried using core show hints , but there are never any hints. Even when a call is actually parked in the correct parking lot. Any tips? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, June 30, 2010 11:27 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking Actually, I should simply have tried. I did need to set CHANNEL(parkinglot). I may have some more questions, but at least it's working right now, and use my own custom extension to pickup the calls. So basically I don't need to (or even can!) include the parking context, I need to setup the extensions myself. For futur reference. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, June 30, 2010 11:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking Hi, Thanks, I thought I could find out about that without installing 1.6, but in the end I did install it on a test server and it answered a few questions. One thing though: I can park calls, in separate private lots, but I can never pick them up again. I have context = some_context defined in features.conf (under the private parking lot) and an include = some_context at the right place, and when I park a call exten = 800 (thats what I use) appears correctly. But I can't seem to pick it up, when I dial 800 it says I am sorry there is no call parked on that extension. This is the relevant context when a call is parked. It clearly shows a call being parked. localhost*CLI dialplan show parkingtest [ Context 'parkingtest' created by 'features' ] '700' = 1. Park() [features] '800' = 1. ParkedCall(800)[features] And the features.conf snippet (everything else is default features.conf from 1.6): [parkinglot_test] context = parkingtest parkpos = 800-805 findslot = next What am I missing you think? I only set the CHANNEL(parkinglot) value when parking the call. Do I need to set that value when picking up a call? (after all, I have no accessz to extension 800 it is created by features.conf) Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aksel Celasun Sent: Monday, June 28, 2010 16:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking Hello there You should have a look at features.conf Regards Aksel Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike Sendt: 28. juni 2010 21:39 Til: 'Asterisk Users Mailing List - Non-Commercial Discussion' Emne: [asterisk-users] Asterisk 1.6 and multiple parking Hi, One of the big features of 1.6 was described as multi-tenant parking. Basically, parking people in different lots so the sales dept. could only pick up their calls, and tech support theirs and no mix up was possible. I can only find the original announcement and others asking the same question. Is there some sort of sample conf file of how I would get this functionnal on the latest 1.6.x? Regards, Mike -- _ -- 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] queue command inasterisk 1.4with macro-argument
I cut and pasted the PHP from your OP and ran it from a shell. When Table AstDB in Database Asterisk contains context foobar, here is the output $php jonas.php VERBOSE query is: SELECT vmcontext FROM AstDB WHERE ID='40' 3 VERBOSE VMCONTEXT is: Array 3 VERBOSE variable exten : 3 VERBOSE variable vmcontext : foobar 3 $ _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, June 30, 2010 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] queue command inasterisk 1.4with macro-argument Danny, 1. I only know php, I'm no programmer 3. the query works in normal PHP. Can I debug to know what's going wrong ? Jonas. On 06/30/2010 05:42 PM, Danny Nicholas wrote: 1. (personal preference) I wouldn't use PHP 2. that out of the way, I comment out the AGI stuff and run my AGI's from bash to make sure the non AGI stuff is happy. 3. the AGI seems to be ok here, I'd make sure my SQL stuff is good. -- _ -- 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 1.6 and multiple parking
I know, I've done this with 1.4 manually with hint extensions. But in 1.6 there is a parameter called parkinghints=yes that is supposed to set them up automatically. It certainly doesn't seem to be doing anything for me. Thanks, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, June 30, 2010 13:38 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking In 1.4 you set up the lots you want to monitor as hints; not sure how this works in 1.6. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, June 30, 2010 11:24 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking Here is my only question left about parkinglots in 1.6. How does the parkinghints=yes parameter work? I've tried using core show hints , but there are never any hints. Even when a call is actually parked in the correct parking lot. Any tips? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, June 30, 2010 11:27 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking Actually, I should simply have tried. I did need to set CHANNEL(parkinglot). I may have some more questions, but at least it's working right now, and use my own custom extension to pickup the calls. So basically I don't need to (or even can!) include the parking context, I need to setup the extensions myself. For futur reference. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, June 30, 2010 11:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking Hi, Thanks, I thought I could find out about that without installing 1.6, but in the end I did install it on a test server and it answered a few questions. One thing though: I can park calls, in separate private lots, but I can never pick them up again. I have context = some_context defined in features.conf (under the private parking lot) and an include = some_context at the right place, and when I park a call exten = 800 (thats what I use) appears correctly. But I can't seem to pick it up, when I dial 800 it says I am sorry there is no call parked on that extension. This is the relevant context when a call is parked. It clearly shows a call being parked. localhost*CLI dialplan show parkingtest [ Context 'parkingtest' created by 'features' ] '700' = 1. Park() [features] '800' = 1. ParkedCall(800)[features] And the features.conf snippet (everything else is default features.conf from 1.6): [parkinglot_test] context = parkingtest parkpos = 800-805 findslot = next What am I missing you think? I only set the CHANNEL(parkinglot) value when parking the call. Do I need to set that value when picking up a call? (after all, I have no accessz to extension 800 it is created by features.conf) Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aksel Celasun Sent: Monday, June 28, 2010 16:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking Hello there You should have a look at features.conf Regards Aksel Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike Sendt: 28. juni 2010 21:39 Til: 'Asterisk Users Mailing List - Non-Commercial Discussion' Emne: [asterisk-users] Asterisk 1.6 and multiple parking Hi, One of the big features of 1.6 was described as multi-tenant parking. Basically, parking people in different lots so the sales dept. could only pick up their calls, and tech support theirs and no mix up was possible. I can only find the original announcement and others asking the same question. Is there some sort of sample conf file of how I would get this functionnal on the latest 1.6.x? Regards, Mike -- _ -- 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] queue command inasterisk 1.4with macro-argument
Thank you for your help. It works now. So these were my first steps into AGI... 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
Re: [asterisk-users] Minimum modules required to run VoIP and CDR
Warren Selby wrote: On Wed, Jun 30, 2010 at 8:50 AM, Frank Church voi...@googlemail.com mailto:voi...@googlemail.com wrote: The DNS setup itself is fine. The sip module just seems to take too much time to load. My modules.conf uses autoload=yes and it seems that many unwanted modules are loaded before sip itself starts. You can stop asterisk and then start it again with asterisk -cvv to see a list of everything that starts up. Take that list, tweak it, then reconfigure your modules.conf to only load the ones you want. Then start it again using the same procedure, making note of any errors that pop up, and resolve them. It will take a little trial and error, but you should be able to get it done. Alternatively, unselect all the modules in menuselect, then just enable the modules as you need them. Leif. -- _ -- 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] Return agi script.
Good afternoon list.
I'm trying to use ${AGISTATUS} after the execution of my script in PHP Agi. But
after running the script, it just returns me 0 (true). Thus:
-- SIP/213-0019AGI Script check.agi completed, returning 0
I tried putting the lines return false; or return 1; but did not change
anything.
Does anyone have a clue?
Thanks,
Rodrigo Lang.
--
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Re: [asterisk-users] Return agi script.
Add void exit (1); to the end of your php script (where you have return 1).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rodrigo Lang
Sent: Wednesday, June 30, 2010 1:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Return agi script.
Good afternoon list.
I'm trying to use ${AGISTATUS} after the execution of my script in PHP Agi.
But after running the script, it just returns me 0 (true). Thus:
-- SIP/213-0019AGI Script check.agi completed, returning 0
I tried putting the lines return false; or return 1; but did not change
anything.
Does anyone have a clue?
Thanks,
Rodrigo Lang.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
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Re: [asterisk-users] Problem with extensions in IVR and queues
Thanks Danny, but I don't know what I should do to fix it... Could you help me? Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010 10:33:31 -0500 Subject: Re: [asterisk-users] Problem with extensions in IVR and queues Sounds like you are getting a “dial without bridge” – asterisk dials x and make the connection, but because the bridge doesn’t happen for what ever reason, the call disconnects like no one ever answered. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Wednesday, June 30, 2010 10:29 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problem with extensions in IVR and queues Hi people, we have some extensions which are included in the IVRs and/or queues. Everything works fine, but the calls done from these extensions are hang up after 30 o 35 seconds. If they are not included in the IVR or queues, the calls are performed well. Do you know if there is something else to set? Thanks, Anahi Ludueña ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! _ Citas sin compromiso por Internet Te damos las claves para encontrar pareja en la red http://contactos.es.msn.com/?mtcmk=015352-- _ -- 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] Return agi script.
It did not work. Returned the broken pipe error. Obs I using phpagi.
Thanks,
Rodrigo Lang.
2010/6/30 Danny Nicholas da...@debsinc.com
Add void exit (1); to the end of your php script (where you have return
1).
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rodrigo Lang
*Sent:* Wednesday, June 30, 2010 1:40 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Return agi script.
Good afternoon list.
I'm trying to use ${AGISTATUS} after the execution of my script in PHP Agi.
But after running the script, it just returns me 0 (true). Thus:
-- SIP/213-0019AGI Script check.agi completed, returning 0
I tried putting the lines return false; or return 1; but did not change
anything.
Does anyone have a clue?
Thanks,
Rodrigo Lang.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Problem with extensions in IVR and queues
Can you post the dialplan section and CLI output from one of these calls? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Wednesday, June 30, 2010 2:05 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problem with extensions in IVR and queues Thanks Danny, but I don't know what I should do to fix it... Could you help me? _ Anahi Ludueña _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010 10:33:31 -0500 Subject: Re: [asterisk-users] Problem with extensions in IVR and queues Sounds like you are getting a dial without bridge asterisk dials x and make the connection, but because the bridge doesnt happen for what ever reason, the call disconnects like no one ever answered. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Wednesday, June 30, 2010 10:29 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problem with extensions in IVR and queues Hi people, we have some extensions which are included in the IVRs and/or queues. Everything works fine, but the calls done from these extensions are hang up after 30 o 35 seconds. If they are not included in the IVR or queues, the calls are performed well. Do you know if there is something else to set? Thanks, _ Anahi Ludueña _ ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! http://www.ayudartepodria.com _ Dime cómo viajas y te diré qué famoso eres ¿Cuál es tu estilo, chic y deslumbrante o mundano y familiar? Descubre quién eres viajando. http://entretenimiento.es.msn.com/test/noticia.aspx?cp-documentid=150990816 -- _ -- 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] Return agi script.
Can you post the script?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rodrigo Lang
Sent: Wednesday, June 30, 2010 2:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Return agi script.
It did not work. Returned the broken pipe error. Obs I using phpagi.
Thanks,
Rodrigo Lang.
2010/6/30 Danny Nicholas da...@debsinc.com
Add void exit (1); to the end of your php script (where you have return 1).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rodrigo Lang
Sent: Wednesday, June 30, 2010 1:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Return agi script.
Good afternoon list.
I'm trying to use ${AGISTATUS} after the execution of my script in PHP Agi.
But after running the script, it just returns me 0 (true). Thus:
-- SIP/213-0019AGI Script check.agi completed, returning 0
I tried putting the lines return false; or return 1; but did not change
anything.
Does anyone have a clue?
Thanks,
Rodrigo Lang.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Return agi script.
Hi Danny. I solve the problem. I put exit (return); where return is equal
to ${AGISTATUS} text. Example:
exit(SUCCESS);
exit(FAILURE);
exit(HANGUP);
This application sets the following channel variable upon completion:
AGISTATUS The status of the attempt to the run the AGI script
text string, one of SUCCESS | FAILURE | NOTFOUND |
HANGUP
:D
Thanks,
Rodrigo Lang.
2010/6/30 Danny Nicholas da...@debsinc.com
Can you post the script?
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rodrigo Lang
*Sent:* Wednesday, June 30, 2010 2:09 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Return agi script.
It did not work. Returned the broken pipe error. Obs I using phpagi.
Thanks,
Rodrigo Lang.
2010/6/30 Danny Nicholas da...@debsinc.com
Add void exit (1); to the end of your php script (where you have return
1).
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rodrigo Lang
*Sent:* Wednesday, June 30, 2010 1:40 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Return agi script.
Good afternoon list.
I'm trying to use ${AGISTATUS} after the execution of my script in PHP Agi.
But after running the script, it just returns me 0 (true). Thus:
-- SIP/213-0019AGI Script check.agi completed, returning 0
I tried putting the lines return false; or return 1; but did not change
anything.
Does anyone have a clue?
Thanks,
Rodrigo Lang.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
_
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Re: [asterisk-users] Problem with extensions in IVR and queues
This is the CLI output, the dialplan is the one that the Elastix creates when somebody sets the followme... I don't know what part you want I post here... Thanks, -- Executing [4...@from-internal:1] GotoIf(SIP/9050-001185aa, 0?ext-local|4010|1) in new stack -- Executing [4...@from-internal:2] Macro(SIP/9050-001185aa, user-callerid|) in new stack -- Executing [...@macro-user-callerid:1] Set(SIP/9050-001185aa, AMPUSER=9050) in new stack -- Executing [...@macro-user-callerid:2] GotoIf(SIP/9050-001185aa, 0?report) in new stack -- Executing [...@macro-user-callerid:3] ExecIf(SIP/9050-001185aa, 1|Set|REALCALLERIDNUM=9050) in new stack -- Executing [...@macro-user-callerid:4] Set(SIP/9050-001185aa, AMPUSER=9050) in new stack -- Executing [...@macro-user-callerid:5] Set(SIP/9050-001185aa, AMPUSERCIDNAME=CALLPBX) in new stack -- Executing [...@macro-user-callerid:6] GotoIf(SIP/9050-001185aa, 0?report) in new stack -- Executing [...@macro-user-callerid:7] Set(SIP/9050-001185aa, AMPUSERCID=9050) in new stack -- Executing [...@macro-user-callerid:8] Set(SIP/9050-001185aa, CALLERID(all)=CALLPBX 9050) in new stack -- Executing [...@macro-user-callerid:9] ExecIf(SIP/9050-001185aa, 0|Set|CHANNEL(language)=) in new stack -- Executing [...@macro-user-callerid:10] GotoIf(SIP/9050-001185aa, 0?continue) in new stack -- Executing [...@macro-user-callerid:11] Set(SIP/9050-001185aa, __TTL=64) in new stack -- Executing [...@macro-user-callerid:12] GotoIf(SIP/9050-001185aa, 1?continue) in new stack -- Goto (macro-user-callerid,s,19) -- Executing [...@macro-user-callerid:19] NoOp(SIP/9050-001185aa, Using CallerID CALLPBX 9050) in new stack -- Executing [4...@from-internal:3] GotoIf(SIP/9050-001185aa, 1?skipdb) in new stack -- Goto (from-internal,4010,5) -- Executing [4...@from-internal:5] Set(SIP/9050-001185aa, __NODEST=) in new stack -- Executing [4...@from-internal:6] Set(SIP/9050-001185aa, __BLKVM_OVERRIDE=BLKVM/4010/SIP/9050-001185aa) in new stack -- Executing [4...@from-internal:7] Set(SIP/9050-001185aa, __BLKVM_BASE=4010) in new stack -- Executing [4...@from-internal:8] Set(SIP/9050-001185aa, DB(BLKVM/4010/SIP/9050-001185aa)=TRUE) in new stack -- Executing [4...@from-internal:9] Set(SIP/9050-001185aa, RRNODEST=) in new stack -- Executing [4...@from-internal:10] Set(SIP/9050-001185aa, __NODEST=4010) in new stack -- Executing [4...@from-internal:11] Set(SIP/9050-001185aa, RecordMethod=Group) in new stack -- Executing [4...@from-internal:12] Macro(SIP/9050-001185aa, record-enable|4010|Group) in new stack -- Executing [...@macro-record-enable:1] GotoIf(SIP/9050-001185aa, 1?check) in new stack -- Goto (macro-record-enable,s,4) -- Executing [...@macro-record-enable:4] AGI(SIP/9050-001185aa, recordingcheck|20100630-154030|1277926830.37214) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck -- AGI Script recordingcheck completed, returning 0 -- Executing [...@macro-record-enable:5] MacroExit(SIP/9050-001185aa, ) in new stack -- Executing [4...@from-internal:13] Set(SIP/9050-001185aa, RingGroupMethod=ringallv2) in new stack -- Executing [4...@from-internal:14] Set(SIP/9050-001185aa, _FMGRP=4010) in new stack -- Executing [4...@from-internal:15] GotoIf(SIP/9050-001185aa, 0?doconfirm) in new stack -- Executing [4...@from-internal:16] Macro(SIP/9050-001185aa, dial|20|tr|4010) in new stack -- Executing [...@macro-dial:1] GotoIf(SIP/9050-001185aa, 1?dial) in new stack -- Goto (macro-dial,s,3) -- Executing [...@macro-dial:3] AGI(SIP/9050-001185aa, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_additional.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 dialparties.agi: Caller ID name is 'CALLPBX' number is '9050' dialparties.agi: USE_CONFIRMATION: 'FALSE' dialparties.agi: RINGGROUP_INDEX: '' dialparties.agi: Methodology of ring is 'ringallv2' -- dialparties.agi: Added extension 4010 to extension map dialparties.agi: got fmgrp_prering: 2, fmgrp_grptime: 20 dialparties.agi: fmgrp_totalprering: 22 dialparties.agi: found extension in pre-ring and array dialparties.agi: ringallv2 ring times: REALPRERING: 22, PRERING: 2 -- dialparties.agi: Extension 4010 cf is disabled -- dialparties.agi: Extension 4010 do not disturb is disabled dialparties.agi: extnum 4010 has: cw: 0; hascfb: 0 [] hascfu: 0 [] dialparties.agi: ExtensionState: 4 dialparties.agi: Extension 4010 has ExtensionState: 4 -- dialparties.agi: Checking CW and CFB status for extension 4010 -- dialparties.agi: dbset CALLTRACE/4010 to 9050
Re: [asterisk-users] Problem with extensions in IVR and queues
Ups, sorry, that CLI output is related to my other problem (the options of IVR doesn't responde when the call is from landline or cell phone). I'll put the correct CLI output... Thanks, Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010 19:50:00 + Subject: Re: [asterisk-users] Problem with extensions in IVR and queues This is the CLI output, the dialplan is the one that the Elastix creates when somebody sets the followme... I don't know what part you want I post here... Thanks, -- Executing [4...@from-internal:1] GotoIf(SIP/9050-001185aa, 0?ext-local|4010|1) in new stack -- Executing [4...@from-internal:2] Macro(SIP/9050-001185aa, user-callerid|) in new stack -- Executing [...@macro-user-callerid:1] Set(SIP/9050-001185aa, AMPUSER=9050) in new stack -- Executing [...@macro-user-callerid:2] GotoIf(SIP/9050-001185aa, 0?report) in new stack -- Executing [...@macro-user-callerid:3] ExecIf(SIP/9050-001185aa, 1|Set|REALCALLERIDNUM=9050) in new stack -- Executing [...@macro-user-callerid:4] Set(SIP/9050-001185aa, AMPUSER=9050) in new stack -- Executing [...@macro-user-callerid:5] Set(SIP/9050-001185aa, AMPUSERCIDNAME=CALLPBX) in new stack -- Executing [...@macro-user-callerid:6] GotoIf(SIP/9050-001185aa, 0?report) in new stack -- Executing [...@macro-user-callerid:7] Set(SIP/9050-001185aa, AMPUSERCID=9050) in new stack -- Executing [...@macro-user-callerid:8] Set(SIP/9050-001185aa, CALLERID(all)=CALLPBX 9050) in new stack -- Executing [...@macro-user-callerid:9] ExecIf(SIP/9050-001185aa, 0|Set|CHANNEL(language)=) in new stack -- Executing [...@macro-user-callerid:10] GotoIf(SIP/9050-001185aa, 0?continue) in new stack -- Executing [...@macro-user-callerid:11] Set(SIP/9050-001185aa, __TTL=64) in new stack -- Executing [...@macro-user-callerid:12] GotoIf(SIP/9050-001185aa, 1?continue) in new stack -- Goto (macro-user-callerid,s,19) -- Executing [...@macro-user-callerid:19] NoOp(SIP/9050-001185aa, Using CallerID CALLPBX 9050) in new stack -- Executing [4...@from-internal:3] GotoIf(SIP/9050-001185aa, 1?skipdb) in new stack -- Goto (from-internal,4010,5) -- Executing [4...@from-internal:5] Set(SIP/9050-001185aa, __NODEST=) in new stack -- Executing [4...@from-internal:6] Set(SIP/9050-001185aa, __BLKVM_OVERRIDE=BLKVM/4010/SIP/9050-001185aa) in new stack -- Executing [4...@from-internal:7] Set(SIP/9050-001185aa, __BLKVM_BASE=4010) in new stack -- Executing [4...@from-internal:8] Set(SIP/9050-001185aa, DB(BLKVM/4010/SIP/9050-001185aa)=TRUE) in new stack -- Executing [4...@from-internal:9] Set(SIP/9050-001185aa, RRNODEST=) in new stack -- Executing [4...@from-internal:10] Set(SIP/9050-001185aa, __NODEST=4010) in new stack -- Executing [4...@from-internal:11] Set(SIP/9050-001185aa, RecordMethod=Group) in new stack -- Executing [4...@from-internal:12] Macro(SIP/9050-001185aa, record-enable|4010|Group) in new stack -- Executing [...@macro-record-enable:1] GotoIf(SIP/9050-001185aa, 1?check) in new stack -- Goto (macro-record-enable,s,4) -- Executing [...@macro-record-enable:4] AGI(SIP/9050-001185aa, recordingcheck|20100630-154030|1277926830.37214) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck -- AGI Script recordingcheck completed, returning 0 -- Executing [...@macro-record-enable:5] MacroExit(SIP/9050-001185aa, ) in new stack -- Executing [4...@from-internal:13] Set(SIP/9050-001185aa, RingGroupMethod=ringallv2) in new stack -- Executing [4...@from-internal:14] Set(SIP/9050-001185aa, _FMGRP=4010) in new stack -- Executing [4...@from-internal:15] GotoIf(SIP/9050-001185aa, 0?doconfirm) in new stack -- Executing [4...@from-internal:16] Macro(SIP/9050-001185aa, dial|20|tr|4010) in new stack -- Executing [...@macro-dial:1] GotoIf(SIP/9050-001185aa, 1?dial) in new stack -- Goto (macro-dial,s,3) -- Executing [...@macro-dial:3] AGI(SIP/9050-001185aa, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_additional.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 dialparties.agi: Caller ID name is 'CALLPBX' number is '9050' dialparties.agi: USE_CONFIRMATION: 'FALSE' dialparties.agi: RINGGROUP_INDEX: '' dialparties.agi: Methodology of ring is 'ringallv2' -- dialparties.agi: Added extension 4010 to extension map dialparties.agi: got fmgrp_prering: 2, fmgrp_grptime: 20 dialparties.agi: fmgrp_totalprering: 22 dialparties.agi: found extension in pre-ring and array dialparties.agi: ringallv2 ring times: REALPRERING: 22, PRERING: 2 -- dialparties.agi
Re: [asterisk-users] Update the LCD with the callee's name after dialing
On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote: Thank you Andrew, I will check it out. We are currently running 1.4. -Matt On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com wrote: Remote Party ID in trunk, it works There are hacks for other versions. We use the patch in https://issues.asterisk.org/view.php?id=6643. Works great. CP -- _ -- 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] Warning spamming for any unsynchronized ISDN port with dahdi-2.3.0.1
We are doing hardware tests with recent dahdi-2.3.0.1 and both asterisk-1.4.33.1 and asterisk-1.6.2.8. Recently, we have noticed that whenever an ISDN port is in RED alarm (unsynchronized), we get a stream of warnings in /var/log/asterisk/full that look like this: [Jun 30 17:38:41] WARNING[9637] chan_dahdi.c: No D-channels available! Using Primary channel 78 as D-channel anyway! [Jun 30 17:38:41] WARNING[9638] chan_dahdi.c: No D-channels available! Using Primary channel 109 as D-channel anyway! [Jun 30 17:38:45] WARNING[9637] chan_dahdi.c: No D-channels available! Using Primary channel 78 as D-channel anyway! [Jun 30 17:38:45] WARNING[9638] chan_dahdi.c: No D-channels available! Using Primary channel 109 as D-channel anyway! [Jun 30 17:38:49] WARNING[9637] chan_dahdi.c: No D-channels available! Using Primary channel 78 as D-channel anyway! [Jun 30 17:38:49] WARNING[9638] chan_dahdi.c: No D-channels available! Using Primary channel 109 as D-channel anyway! [Jun 30 17:38:50] VERBOSE[9626] asterisk.c: -- Remote UNIX connection [Jun 30 17:38:53] WARNING[9637] chan_dahdi.c: No D-channels available! Using Primary channel 78 as D-channel anyway! [Jun 30 17:38:53] WARNING[9638] chan_dahdi.c: No D-channels available! Using Primary channel 109 as D-channel anyway! [Jun 30 17:38:57] WARNING[9637] chan_dahdi.c: No D-channels available! Using Primary channel 78 as D-channel anyway! [Jun 30 17:38:57] WARNING[9638] chan_dahdi.c: No D-channels available! Using Primary channel 109 as D-channel anyway! This particular machine runs asterisk-1.6.2.8 with dahdi-2.3.0.1. The telephony hardware is an OpenVox PRI card with four E1 ports (driver is wct4xxp). On this system, I have configured port 1 as pri_net, and port 2 as pri_cpe, and connected the two with a crossover cable, leaving ports 3 and 4 disconnected. Therefore I have two synchronized ports (with each other) and two unsynchronized ports (RED). From what I see in /proc/dahdi/* , channels 78 and 109 are the two D channels of the two disconnected ports. I can route calls between the two connected ports, so that part appears to work OK. I have reproduced this stream of warnings on another machine with asterisk-1.4.33.1 and dahdi-2.3.0.1, and also with other card types (OpenVox with 1 E1 port, Sangoma with 2 T1 ports, Rhino with 2 T1 ports), so I do not think the particular driver is an issue. The question I have is this: is this warning message something to be expected from ports with RED alarms? Or is this message a symptom of a deeper misconfiguration? Since I am the package manager for the Elastix project (http://www.elastix.org), I am the one who can solve misconfigurations, if any. -- _ -- 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] Pbx_lua vs. calling lua thru AGI?
Hello I'm taking a look at how to write scripts to be called from the dialplan, and saw pbx_lua mentioned. I'd like to know more about this feature, such as what the difference is with just calling the Lua interpreter through AGI (same difference as between php-cgi and mod_php?), whether it's production-ready, etc. Thank you for any help. -- _ -- 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] [voice mail] Estimating file size?
On Sat, 26 Jun 2010 17:53:27 +0100 (BST), Gordon Henderson gordon+aster...@drogon.net wrote: Dial an extension that answers and stores to voicemail, say blah blah into it for one minute and check the resulting file size. divide it by 60 and you'll get a good estimate of the number of bytes per second of recording for your chosen format. Thanks for the tip. I'll give it a try and see how much space VM msgs take. -- _ -- 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 stop intruder from registering sip?
On Sun, 13 Jun 2010, Tilghman Lesher wrote: I would generally suggest something a little more deterministic (where 101 is your extension): $ echo '101This is a salt' | sha1sum 22c3c098bfc2289396af84ecfb1ab77419a6537e Aside from being 8 characters longer, why do you prefer sha1sum to md5sum? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Pbx_lua vs. calling lua thru AGI?
On Thu, 1 Jul 2010, Gilles wrote: I'm taking a look at how to write scripts to be called from the dialplan, and saw pbx_lua mentioned. I'd like to know more about this feature, such as what the difference is with just calling the Lua interpreter through AGI (same difference as between php-cgi and mod_php?), whether it's production-ready, etc. I've never used it (I'm a 1.2 Luddite), but I would be very interested in anything that looks like a real language for writing dialplans. I've used AEL, and it is a much cleaner method of writing dialplan, but (at least in 1.2) it has a bunch of syntactical weirdness. For example, forgetting to end a line with a semi-colon can cause half your dialplan to disappear without warning. I use AGI a lot. I write AGIs in C so, aside from the create a new process hit, they execute at the same speed as the code inside Asterisk. You can execute XXX AGIs written in C in the time it takes to load either the Perl or PHP interpreter and parse your script. AGIs have a lot of advantages: ) If they crash, they only impact the call that invoked the AGI. ) They're nice little black boxes where you can package up a bunch of logic and complexity in a single line of dialplan. Imagine implementing voicemail in dialplan versus an AGI. AGIs hide a lot of detail and help keep clumsy fingers from introducing impossible to find bugs. ) They can be debugged separately from Asterisk. By feeding the appropriate input (by file redirection) from the command line, you can do a substantial bit of debugging as long as it doesn't need to actually interact with Asterisk. I frequently fire up emacs, load gdb, and step through my C AGIs line by line. Set a breakpoint, examine a variable, change it's value, and continue. Sure beats the heck out of trying to debug an AGI by peppering it with VERBOSE or syslog() statements. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 the LCD with the callee's name after dialing
On Wed, Jun 30, 2010 at 6:10 PM, CunningPike cunningp...@gmail.com wrote: On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote: Thank you Andrew, I will check it out. We are currently running 1.4. -Matt On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com wrote: Remote Party ID in trunk, it works There are hacks for other versions. We use the patch in https://issues.asterisk.org/view.php?id=6643. Works great. CP Until Asterisk 1.8 is released this looks like the easiest way to get remote party id working. I have modified the patch to work with Asterisk 1.6.2.9. I have also attached a patch against FreePBX 2.7 to add the necessary changes to the dialplan. I have verified this works on a Polycom 550. Ryan asterisk-1.6.2.9-called-rpid.patch Description: Binary data freepbx-2.7.0.8-core-called-rpid.patch Description: Binary data -- _ -- 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] Want to retrieve the value of contact header
Dear all, I want to retrieve the value from Contact header and from From header which is 0345001280 from the following two lines: Contact: sip:0345001...@123.50.217.143 sip%3a0345001...@123.50.217.143 From: 99 sip:0345001...@113.34.235.106sip%3a0345001...@113.34.235.106 ;tag=as191896a1 Is it possible in asterisk to retrieve that value? I am getting following value in the corresponding variable when I pass the following SIP message. Is there anything which contain the value of 0345001280 of contact ? Corresponding value: CALLERID(num): 185475 CALLERID(name) : 99 SCI-PEERNAME : 185475 SIP message: INVITE sip:08058913...@113.34.235.106 sip%3a08058913...@113.34.235.106SIP/2.0 Via: SIP/2.0/UDP 123.50.217.143:5060;branch=z9hG4bK100b063a;rport From: 99 sip:0345001...@113.34.235.106sip%3a0345001...@113.34.235.106 ;tag=as191896a1 To: sip:08058913...@113.34.235.106 sip%3a08058913...@113.34.235.106 Contact: sip:0345001...@123.50.217.143 sip%3a0345001...@123.50.217.143 Call-ID: 0f3fbfe3463035d04f05534824a18...@113.34.235.106 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 01 Jul 2010 02:20:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 267 v=0 o=root 22702 22702 IN IP4 123.50.217.143 s=session c=IN IP4 123.50.217.143 t=0 0 m=audio 17262 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - Is it possible to retrieve the value of contact in asterisk ? Please let me know. Is there anyone who knows the solution? I need this urgent. Thanks in advance Nahar -- _ -- 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] Want to retrieve the value of contact header
You might take a look at the SIPHEADER function which can return specific SIP headers. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 30, 2010, at 7:36 PM, kamrun nahar bina wrote: Dear all, I want to retrieve the value from Contact header and from From header which is 0345001280 from the following two lines: Contact: sip:0345001...@123.50.217.143 From: 99 sip:0345001...@113.34.235.106;tag=as191896a1 Is it possible in asterisk to retrieve that value? I am getting following value in the corresponding variable when I pass the following SIP message. Is there anything which contain the value of 0345001280 of contact ? Corresponding value: CALLERID(num): 185475 CALLERID(name) : 99 SCI-PEERNAME : 185475 SIP message: INVITE sip:08058913...@113.34.235.106 SIP/2.0 Via: SIP/2.0/UDP 123.50.217.143:5060;branch=z9hG4bK100b063a;rport From: 99 sip:0345001...@113.34.235.106;tag=as191896a1 To: sip:08058913...@113.34.235.106 Contact: sip:0345001...@123.50.217.143 Call-ID: 0f3fbfe3463035d04f05534824a18...@113.34.235.106 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 01 Jul 2010 02:20:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 267 v=0 o=root 22702 22702 IN IP4 123.50.217.143 s=session c=IN IP4 123.50.217.143 t=0 0 m=audio 17262 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - Is it possible to retrieve the value of contact in asterisk ? Please let me know. Is there anyone who knows the solution? I need this urgent. Thanks in advance Nahar -- _ -- 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] call file question
I am sure this is simple, but have been struggling. I want to create a
call file that dials out a particular Dahdi channel to enable call
forwarding on a POTS line. I have this in extensions.conf:
[custom-callfwd]
exten = s,1,Answer
exten = s,n,Dial(DAHDI/4-1/*717157750)
exten = s,n,Verbose(${DIALSTATUS})
exten = s,n,Hangup
[custom-callfwdcanc]
exten = s,1,Answer
exten = s,n,Dial(DAHDI/4-1/*72)
exten = s,n,Verbose(${DIALSTATUS})
exten = s,n,Hangup
Using FreePBX I have setup custom destinations and custom
applications such that users can dial a code from their desks and enable
or disable forwarding via the above contexts. That works fine.
Now I whipped up a C program to create a call file to do the same thing
from the command line:
[snip]
fprintf(callfile, Channel: Local/*...@custom-callfwd/n\n);
fprintf(callfile, Application: Playback\n);
fprintf(callfile, Data: hello-world\n);
[snip]
When I run this it creates the call file and I see this in the console:
-- Attempting call on Local/*...@custom-callfwd/n for application
Playback(hello-world) (Retry 1)
And that is all... no call actually goes out on the Dahdi line.
I'm sure I am not properly creating the call file to do what I want. Any
suggestions?
Thanks,
j
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Re: [asterisk-users] call file question
On Thu, 1 Jul 2010, Jeff LaCoursiere wrote:
I am sure this is simple, but have been struggling. I want to create a
call file that dials out a particular Dahdi channel to enable call
forwarding on a POTS line. I have this in extensions.conf:
[custom-callfwd]
exten = s,1,Answer
exten = s,n,Dial(DAHDI/4-1/*717157750)
exten = s,n,Verbose(${DIALSTATUS})
exten = s,n,Hangup
[custom-callfwdcanc]
exten = s,1,Answer
exten = s,n,Dial(DAHDI/4-1/*72)
exten = s,n,Verbose(${DIALSTATUS})
exten = s,n,Hangup
Using FreePBX I have setup custom destinations and custom
applications such that users can dial a code from their desks and
enable or disable forwarding via the above contexts. That works fine.
Now I whipped up a C program to create a call file to do the same thing
from the command line:
[snip]
fprintf(callfile, Channel: Local/*...@custom-callfwd/n\n);
I don't see exten *71 in custom-callfwd.
Why are you using a local channel in your call file?
fprintf(callfile, Application: Playback\n);
fprintf(callfile, Data: hello-world\n);
[snip]
When I run this it creates the call file and I see this in the console:
-- Attempting call on Local/*...@custom-callfwd/n for application
Playback(hello-world) (Retry 1)
What does the call file look like before you mv it to the spool directory?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] How to stop intruder from registering sip?
On Wednesday 30 June 2010 18:38:51 Steve Edwards wrote: On Sun, 13 Jun 2010, Tilghman Lesher wrote: I would generally suggest something a little more deterministic (where 101 is your extension): $ echo '101This is a salt' | sha1sum 22c3c098bfc2289396af84ecfb1ab77419a6537e Aside from being 8 characters longer, why do you prefer sha1sum to md5sum? The use of MD5 is gradually being displaced, as crypto attacks are getting better. Since SHA1 is usually the replacement, I went with it, since it's also likely to be available on systems. While SHA1 will eventually succumb to the same attacks as MD5, due to its larger bitstrength, it has quite a few years left in it, before we need to start thinking about SHA256 or SHA512 to replace it. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.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] Return agi script.
On Wednesday 30 June 2010 13:39:57 Rodrigo Lang wrote:
Good afternoon list.
I'm trying to use ${AGISTATUS} after the execution of my script in PHP Agi.
But after running the script, it just returns me 0 (true). Thus:
-- SIP/213-0019AGI Script check.agi completed, returning 0
I tried putting the lines return false; or return 1; but did not change
anything.
Does anyone have a clue?
If you want to set a value, use the SET VARIABLE agi command to do so.
The AGISTATUS variable tells you nothing more than whether your script ran,
failed to run (not executable), could not be found (typo!), or exited because
the calling channel hung up.
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Re: [asterisk-users] Want to retrieve the value of contact header
Dear Jim Dickenson. Ｔｈａｎｋｓ for you mail. I have got the solution. Thanks Nahar On Thu, Jul 1, 2010 at 11:45 AM, Jim Dickenson dicken...@cfmc.com wrote: You might take a look at the SIPHEADER function which can return specific SIP headers. -- Jim Dickenson mailto:dicken...@cfmc.com dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 30, 2010, at 7:36 PM, kamrun nahar bina wrote: Dear all, I want to retrieve the value from Contact header and from From header which is 0345001280 from the following two lines: Contact: sip:0345001...@123.50.217.143 sip%3a0345001...@123.50.217.143 From: 99 sip:0345001...@113.34.235.106sip%3a0345001...@113.34.235.106 ;tag=as191896a1 Is it possible in asterisk to retrieve that value? I am getting following value in the corresponding variable when I pass the following SIP message. Is there anything which contain the value of 0345001280 of contact ? Corresponding value: CALLERID(num): 185475 CALLERID(name) : 99 SCI-PEERNAME : 185475 SIP message: INVITE sip:08058913...@113.34.235.106 sip%3a08058913...@113.34.235.106SIP/2.0 Via: SIP/2.0/UDP 123.50.217.143:5060;branch=z9hG4bK100b063a;rport From: 99 sip:0345001...@113.34.235.106sip%3a0345001...@113.34.235.106 ;tag=as191896a1 To: sip:08058913...@113.34.235.106 sip%3a08058913...@113.34.235.106 Contact: sip:0345001...@123.50.217.143 sip%3a0345001...@123.50.217.143 Call-ID: 0f3fbfe3463035d04f05534824a18...@113.34.235.106 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 01 Jul 2010 02:20:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 267 v=0 o=root 22702 22702 IN IP4 123.50.217.143 s=session c=IN IP4 123.50.217.143 t=0 0 m=audio 17262 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - Is it possible to retrieve the value of contact in asterisk ? Please let me know. Is there anyone who knows the solution? I need this urgent. Thanks in advance Nahar -- _ -- 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
