search for: michofr

Displaying 20 results from an estimated 52 matches for "michofr".

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2009 Jan 19
6
G729 codec
Dear All, I have the following CPU info on my asterisk server: Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST 2008 i686 i686 i386 GNU/Linux I need to install G729 on the asterisk server just to pass through and not for encoding...Which G729 package do you advice me to install? I tried several packages with no luck Regards -------------- next part -------------- An
2008 Sep 12
1
Extension not found
...lling] exten => _X.,1,Gotoif($[${EXTEN} = 111] ? custom-recordme,111,1) exten => _X.,2,DeadAGI,a2billing.php exten => _X.,3,Wait,2 exten => _X.,4,Hangup But i have the following error when trying to dial 111: [Sep 12 14:16:32] WARNING[30978]: pbx.c:2483 __ast_pbx_run: Channel 'SIP/michofr-093833e0' sent into invalid extension '111' in context ' custom-recordme', but no invalid handler Any help? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080912/c202786a/at...
2010 Jun 19
2
Muti Asterisk
Dear All, I have installed 4 asterisks on the same Centos machine..>Each Asterisk has its own installation folder and use its own libraries...Everything looks great and all asterisks are doing their jobs correctly except one thing...I faced a voice quality issue...On a specific time, and after the number of calls begin increasing, the voice quality will begin degradation... Could it be a
2009 Feb 28
2
No rtp activity
Hi all.... I'm using asterisk for making PSTN calls from extensions registered on OpenSIPS...In peak hours ,number of calls Increase dramatically to a non logic number..When checking the calls using asterisk CLI I saw a lot of calls in ringing status and after 300s(rtphold timeout), asterisk release all calls...I checked the log file and found.. [Feb 28 11:34:14] NOTICE[19197] chan_sip.c:
2008 Sep 23
5
Extension registration
Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards -------------- next part
2009 Feb 17
4
Network architecture
Hi all, I'm planning to build a VOIP solution for handling SIP calls coming from endpoints registered on a specific SIP proxy...I made some research regarding network architecture and found out that the best solution is to use OpenSips as SIP proxy for registration and local calls between registered endpoints and use asterisk server with a2billing for PSTN calls, rating, routing and all other
2008 Nov 20
1
Voice Mail
...wing configuration: * extensions.conf:* exten => _999.,1,VoiceMail(${EXTEN}) exten => _999.,2,HangUp() If the customer dial 9991234 then a prompt message should ask him to enter his voice message and this what is not happening *voicemail.conf:* [a2billing] 999123456 => 123456, 123456, michofr at mm.com The Error mrssage that I got is: Executing [999123456 at a2billing:1] VoiceMail("SIP/a555555-0822cfa8", "999123456") in new stack [Nov 20 10:12:09] WARNING[26599]: app_voicemail.c:2862 leave_voicemail: No entry in voicemail config file for '999123456' --...
2008 Dec 22
1
No Audio
Hi all, Sometimes when making a PC to PSTN call through asterisk, I got no audio in both sides...tracing by wireshark, I can find that RTP packets are hitting my PC but no audio...Can someone guess what could be that issue? Maybe it's a latency issue? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:
2009 Feb 24
1
Incoming call
Dera All, I have the following scenario, A customer dial a DID number...The call is routed to a PSTN GW that send the call to asterisk... On asterisk I created an AGI Script that send the call to an extension registered on OpenSIPS server... The extension is ringing successfully, but as soon as I accept the call on OpenSIPS side the call is hangd up... I checked rhe SIP debug and it seems that I
2009 Feb 26
1
incoming call problem
Dear All, I have created an inbound context in SIP .conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl = yes...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contains T.38 protocol...When t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with OpenSIPS and cal
2009 Mar 02
1
Asterisk Dial plan issue
Hi all, I'm using asterisk in real time mode...My extensions.conf table contains: [default] switch => Realtime/default at extensions I have added the following to extensions.conf table; context:micho exten: _X. priority: 1 app:Dial appdata: SIP/00XXXXXX at PSTN GAteway Asterisk server is connected succeffully to database...As soon as i make a call i got the following error message:
2009 Mar 20
1
T38 FAX
Dear All, I'm trying to send FAX to an endpoint Behind NAT...The scenario i the following: PSTN_GW-->Asterisk-->asterisk-->OpenSIPS-->Endpoint behind NAT.. The FAX is failed and I got the following error log on asterisk: Mar 20 09:21:09] WARNING[20388]: chan_sip.c:12409 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog
2009 Apr 21
1
run dialplan when open line
Hi all, Does asterisk support the following scenario? I need when a customer who own an endpoint registered on asterisk open the line, the asterisk will run a specific AGI script inside the endpoint context? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090421/4dce9c34/attachment.htm
2010 Aug 05
1
Codec Conversion
Dear All, i would like to ask please if someone tried to make a codec conversion from ilbc to g729, because i did that but the voice quality was too bad and a lot of disconnection.. Can i get your feedback regarding this issue please? regards -------------- next part -------------- An HTML attachment was scrubbed... URL:
2009 Mar 02
2
Asterisk realtime
Hi all, I'm using asterisk in real time mode...All extensions are defined in table sip_buddies...Everything looks fine and asterisk is reading extensions info from the sip_buddies table...The problem occurs as soon as any information on an extension is changed from sip_buddies table...Which mean, if I change the secret field in sip_buddies table then i should reload asterisk to read again the
2009 Jan 27
2
T.38
Dear All, I'm trying to send Fax using T.38 protocol but the FAX is not going through..I'm getting the following error om /var/log/messages [Jan 27 16:46:31] WARNING[25435] channel.c: No path to translate from SIP/80.169.210.181-0896bfd0(4) to SIP/028949469-b7703d40(256) [Jan 27 16:46:40] WARNING[24349] channel.c: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw)
2009 Feb 19
3
AGI script
Dear All, I would like to ask please if someone has a AGI script that select a value from a database and dial this value as a destination number Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090220/e2aa530c/attachment.htm
2009 Feb 11
2
OPTIONS packets
Hi all, I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy is not replying back...The issue is the UNKNOWN username that reside in the OPTIONS packet as you can see in the captured packets as you can see below: 1. U Asterisk_IP:5060 -> OPENSIPS_IP:5060 2. OPTIONS sip:OPENSIPS_IP
2008 Dec 15
3
tcpdum
*Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 > Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean
2009 Jan 16
0
No subject
...address'<o:p></o:p></span></font></p> <div> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span = style=3D'font-size: 12.0pt'>On Thu, Jan 29, 2009 at 9:18 PM, michel freiha &lt;<a href=3D"mailto:michofr at gmail.com">michofr at gmail.com</a>&gt; = wrote:<o:p></o:p></span></font></p> <div> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span = style=3D'font-size: 12.0pt'>Do you mean call limit...