Re: [asterisk-users] Common/Reasonable Assumption on DID/Channel over-subscription
I suspected as much :) Well, it IS a calling card; people call an access number, dial an international number. Assuming typical ALOC to be 8 mins which is seen quite often in International calls esp. in ethnic communities, and since the service hasn't launched yet, it's hard to tell what the incoming traffic will be like but in order for us to purchase the channel packs, we do need to figure out the ratio of over-subscription we can use for the number of channels to buy so while I understand it's a little vague, just wanted to hear from people who're running similar services and what is their actual channel usage and if they have consciously designed it using an assumption for this ratio or they just buy more channels and/or DIDs looking at historical data (or customer complaints) On Sat, May 26, 2012 at 8:46 PM, Don Kelly d...@donkelly.biz wrote: I don’t think it’s possible to suggest a ratio without knowing what your actual application “similar to calling card services” is. --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *A E [Gmail] *Sent:* Saturday, May 26, 2012 5:13 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion; FreeSWITCH Users Help *Subject:* [asterisk-users] Common/Reasonable Assumption on DID/Channel over-subscription ** ** Hello All, ** ** just throwing this out there. What are people generally using these days when designing their services, esp. those that require a user to call a DID to access their system, similar to calling card services. There was a time when this used to be 50 to 1 for DIDs, and about 10 to 1 for number of channels bought in SMB with IP-PBX. ** ** I believe this would have changed today and assuming a service is pretty popular, the ALOCs are longer due to cheaper rates and convenience of calling. Does anyone have any real world numbers they can share? Is 10 to 1 a good ratio to ensure a user practically never gets a circuits are busy? ** ** Thanks in advance -- _ -- 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] Deleting OLD Voicemails
I believe one of the patches involved in fixing for The Great Voicemail Problem* about a year ago was to make voicemail automatically renumber the mailbox files if it saw a gap. * from memory: The Great Voicemail Problem is a bug where if you received a new voicemail while listening to a message, the mailbox was not renumbered correctly when you deleted a message. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias Sent: Saturday, May 26, 2012 10:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Deleting OLD Voicemails I did not understand. What do you mean with renumber all the messages? El 25/05/2012 02:27, Edwin Lam edwin@officegeneral.com escribió: On 5/23/12 2:42 AM, Danny Dias wrote: Can i delete like this: rm -rf /var/spool/asterisk/voicemail/voicemailcontextcustomer/300/INBOX/*.* Is that ok? will this break something? that's ok no it shouldn't break anything. however if you're going to delete some of the messages. you have to renumber all the messages so that they are consecutive otherwise the voicemail application may give you grief. A little doubt here, once the user hears the voicemail using the phone, the message is automatically moved to Old folder, is that right? yes -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 tel:%2B1%20415%20439%204988 Fax: +1 415 283 3370 tel:%2B1%20415%20283%203370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- 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 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 Forwarding
Hi Guys, Seeing an issue with 1.6.2.17.2 and also 1.6.2.14 When we do call forwarding if the call coming in to be forwarded asterisk sends the invite out to our ITSP as username@anonymous.invalid instead of username@domain. When call comes in with CLI and is forwarded it sends it as username@domain to our ITSP. Is this a bug or is there something I need to turn on or off? All the ITSP's we use authenticate on username and domain. Thanks Brian. -- _ -- 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] Deleting OLD Voicemails
I use find on a cron schedule to remove old recordings everyday. Im sure
you can do the same
find -H /var/log/asterisk/asterisk_rec/* -mtime +90 -type f -exec rm -v {}
\;
anything older than 90 days
On 27 May 2012 09:20, Eric Wieling ewiel...@nyigc.com wrote:
I believe one of the patches involved in fixing for The Great Voicemail
Problem* about a year ago was to make voicemail automatically renumber the
mailbox files if it saw a gap.
* from memory: The Great Voicemail Problem is a bug where if you received
a new voicemail while listening to a message, the mailbox was not
renumbered correctly when you deleted a message.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias
Sent: Saturday, May 26, 2012 10:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Deleting OLD Voicemails
I did not understand. What do you mean with renumber all the messages?
El 25/05/2012 02:27, Edwin Lam edwin@officegeneral.com escribió:
On 5/23/12 2:42 AM, Danny Dias wrote:
Can i delete like this:
rm -rf
/var/spool/asterisk/voicemail/voicemailcontextcustomer/300/INBOX/*.*
Is that ok? will this break something?
that's ok
no it shouldn't break anything.
however if you're going to delete some of the messages. you have to
renumber all the messages so that they are consecutive otherwise
the voicemail application may give you grief.
A little doubt here, once the user hears the voicemail
using the phone, the
message is automatically moved to Old folder, is that right?
yes
--
Edwin Lam edwin@officegeneral.com
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 tel:%2B1%20415%20439%204988 Fax: +1 415
283 3370 tel:%2B1%20415%20283%203370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
--
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[asterisk-users] NAT problem: Retransmission timeout reached on transmission … for seqno 2 (Critical Response)
I can't receive an incoming call from a DID provider to my NATted Asterisk box. I'm testing this by dialling my DID with Skype, since I can't dial it from my mobile phone (as it's an iNum). I specified the public IP to Asterisk using externhost but also tried externip, and it didn't help. I can receive calls directly over SIP that don't use my DID. The phone rings, but the call won't complete, and the error I get is: Retransmission timeout reached on transmission MDVkZWU1YzcxNTBhNzU0OTZhNDJjODMxMGM4ZTBmMmI. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response Here is what I get when I turn on SIP debugging (the Asterisk box's private IP is shown as 192.168.15.200, its external IP is shown as 60.70.80.90, the handset that Asterisk rings is shown as 192.168.15.122, the other IPs are presumably from the DID provider and/or Skype): --- SIP read from UDP:212.8.163.67:5061 --- INVITE sip:883510001288...@servalan.malcolm.id.au SIP/2.0 Record-Route: sip:212.8.163.67:5061;r2=on;lr;ftag=5ba33723 Record-Route: sip:192.168.34.151:5061;r2=on;lr;ftag=5ba33723 Via: SIP/2.0/UDP 212.8.163.67:5061;branch=z9hG4bK-d8754z-b37c8438ea03411d-1---d8754z- Via: SIP/2.0/UDP 192.168.34.202:16000;received=192.168.34.202;branch=z9hG4bK-d8754z-b37c8438ea03411d-1---d8754z-;rport=16000 Max-Forwards: 70 Contact: sip:123456@192.168.34.202:16000 To: sip:883510001288388@60.70.80.90:5060 From: skypeusernamesip:123456@192.168.34.202;tag=5ba33723 Call-ID: MzJiYmI0M2RmNThmNWM2NDk3OWY0OGVmNjFkNTJkNGI. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Type: application/sdp User-Agent: SipGW 8 Privacy: id P-Asserted-Identity: skypeusernamesip:123456@192.168.34.202 Remote-Party-ID: skypeusernamesip:123456@192.168.34.202;party=calling;screen=yes;privacy=full Content-Length: 463 v=0 o=123456 1338117946 1338117946 IN IP4 213.19.129.6 s=Skype call c=IN IP4 213.19.129.6 t=0 0 m=audio 35336 RTP/AVP 18 0 8 104 102 103 117 116 124 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:104 SILK_WB_V3/16000 a=rtpmap:102 SILK_MB_V3/12000 a=rtpmap:103 SILK_NB_V3/8000 a=rtpmap:117 NWC/16000 a=rtpmap:116 UNCODEDWB/16000 a=rtpmap:124 UNCODEDSWB/24000 a=rtpmap:101 telephone-event/8000 - --- (18 headers 17 lines) --- Sending to 212.8.163.67:5061 (NAT) Using INVITE request as basis request - MzJiYmI0M2RmNThmNWM2NDk3OWY0OGVmNjFkNTJkNGI. No matching peer for '123456' from '212.8.163.67:5061' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 104 Found RTP audio format 102 Found RTP audio format 103 Found RTP audio format 117 Found RTP audio format 116 Found RTP audio format 124 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found unknown media description format SILK_WB_V3 for ID 104 Found unknown media description format SILK_MB_V3 for ID 102 Found unknown media description format SILK_NB_V3 for ID 103 Found unknown media description format NWC for ID 117 Found unknown media description format UNCODEDWB for ID 116 Found unknown media description format UNCODEDSWB for ID 124 Found audio description format telephone-event for ID 101 Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 213.19.129.6:35336 Looking for 883510001288388 in default (domain servalan.malcolm.id.au) list_route: hop: sip:212.8.163.67:5061;r2=on;lr;ftag=5ba33723 list_route: hop: sip:192.168.34.151:5061;r2=on;lr;ftag=5ba33723 --- Transmitting (NAT) to 212.8.163.67:5061 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.8.163.67:5061;branch=z9hG4bK-d8754z-b37c8438ea03411d-1---d8754z-;received=212.8.163.67;rport=5061 Via: SIP/2.0/UDP 192.168.34.202:16000;received=192.168.34.202;branch=z9hG4bK-d8754z-b37c8438ea03411d-1---d8754z-;rport=16000 Record-Route: sip:212.8.163.67:5061;r2=on;lr;ftag=5ba33723 Record-Route: sip:192.168.34.151:5061;r2=on;lr;ftag=5ba33723 From: skypeusernamesip:123456@192.168.34.202;tag=5ba33723 To: sip:883510001288388@60.70.80.90:5060 Call-ID: MzJiYmI0M2RmNThmNWM2NDk3OWY0OGVmNjFkNTJkNGI. CSeq: 2 INVITE Server: Asterisk PBX 1.8.11.1~dfsg-1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:883510001288388@118.107.224.38:5060 Content-Length: 0 Audio is at 10226 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.15.122:5060: INVITE sip:asteriskuser@192.168.15.122:5060;transport=udp SIP/2.0
Re: [asterisk-users] Common/Reasonable Assumption on DID/Channel over-subscription
The users list probably isn't the best place for this discussion. Send me a note directly if you like. --Don From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail] Sent: Sunday, May 27, 2012 1:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Common/Reasonable Assumption on DID/Channel over-subscription I suspected as much :) Well, it IS a calling card; people call an access number, dial an international number. Assuming typical ALOC to be 8 mins which is seen quite often in International calls esp. in ethnic communities, and since the service hasn't launched yet, it's hard to tell what the incoming traffic will be like but in order for us to purchase the channel packs, we do need to figure out the ratio of over-subscription we can use for the number of channels to buy so while I understand it's a little vague, just wanted to hear from people who're running similar services and what is their actual channel usage and if they have consciously designed it using an assumption for this ratio or they just buy more channels and/or DIDs looking at historical data (or customer complaints) On Sat, May 26, 2012 at 8:46 PM, Don Kelly d...@donkelly.biz wrote: I don't think it's possible to suggest a ratio without knowing what your actual application similar to calling card services is. --Don Don Kelly PCF Corp People Come First 651 842-1000 tel:651%20842-1000 651 842-1001 tel:651%20842-1001 fax From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail] Sent: Saturday, May 26, 2012 5:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; FreeSWITCH Users Help Subject: [asterisk-users] Common/Reasonable Assumption on DID/Channel over-subscription Hello All, just throwing this out there. What are people generally using these days when designing their services, esp. those that require a user to call a DID to access their system, similar to calling card services. There was a time when this used to be 50 to 1 for DIDs, and about 10 to 1 for number of channels bought in SMB with IP-PBX. I believe this would have changed today and assuming a service is pretty popular, the ALOCs are longer due to cheaper rates and convenience of calling. Does anyone have any real world numbers they can share? Is 10 to 1 a good ratio to ensure a user practically never gets a circuits are busy? Thanks in advance -- _ -- 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] Telephony Card: GSM slots + Analoge
On Sat, May 26, 2012 at 9:52 AM, Moises Silva moises.si...@gmail.comwrote: There is nothing hybrid like that (GSM + Analog) in the NorthAmerica or Europe to my knowledge. We at Sangoma (from Canada) have a 4-port GSM card though which uses chan_dahdi (patching needed at the moment). Actually Beronet (Germany) manufacture such a hybrid solution - you have to check berofix line (http://www.berofix.com/). Not many options (like 2xGSM + 4FXO or 2xGSM + 4FXS per card) but maybe it is what you need. HTH, Ioan -- _ -- 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] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
Hi list, we are upgrading our Asterisk production server from 1.6.24 to 1.8.12 version and face the following problem: one of our peer (voicetrading.com) doesn't accept our calls anymore, we receive a timeout error Packet timed out after 32000ms with no response. Switching back to 1.6 make things working again! In sip.conf we have nat=no, peer conf is: [myPeerDef] type=peer host=111.111.1.111 context=honeypot insecure=invite directmedia=no disallow=all allow=ulaw,alaw dtmfmode=inband We aren't registered, our IP is authorized by their system. Debug of sessions (222.222.22.22 is our server 111.111.1.111 is their) Working one with 1.6: Audio is at 222.222.22.22 port 26002 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 111.111.1.111:5060: INVITE sip:0336@111.111.1.111 SIP/2.0 Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport Max-Forwards: 70 From: TOOTAi sip:00333@222.222.22.22;tag=as52190c5c To: sip:0336@111.111.1.111 Contact: sip:00333@222.222.22.22 Call-ID: 2c974a0a2b08abe320ed388433e47d7e@222.222.22.22 CSeq: 102 INVITE User-Agent: TOOTAiAudio Date: Sun, 27 May 2012 16:10:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 199 v=0 o=root 284043376 284043376 IN IP4 222.222.22.22 s=TOOTAiAudio PBX c=IN IP4 222.222.22.22 t=0 0 m=audio 26002 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv --- -- Called myPeerDef/0336 --- SIP read from UDP:111.111.1.111:5060 --- SIP/2.0 183 Session progress Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport From: TOOTAi sip:00333@222.222.22.22;tag=as52190c5c To: sip:0336@111.111.1.111;tag=4e0313ac670313ac4f9920c3173f554 Contact: sip:0336@111.111.1.111:5060 Call-ID: 2c974a0a2b08abe320ed388433e47d7e@222.222.22.22 CSeq: 102 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 159 v=0 o=CARRIER 1338135040 1338135040 IN IP4 77.72.168.74 s=SIP Call c=IN IP4 77.72.168.74 t=0 0 m=audio 18456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 - --- (11 headers 8 lines) --- Found RTP audio format 0 Found audio description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 77.72.168.74:18456 Peer doesn't provide video -- SIP/myPeerDef-0007 is making progress passing it to SIP/104-0006 --- SIP read from UDP:111.111.1.111:5060 --- SIP/2.0 200 Ok Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport From: TOOTAi sip:00333@222.222.22.22;tag=as52190c5c To: sip:0336@111.111.1.111;tag=4e0313ac670313ac4f9920c3173f554 Contact: sip:0336@111.111.1.111:5060 Call-ID: 2c974a0a2b08abe320ed388433e47d7e@222.222.22.22 CSeq: 102 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 159 v=0 o=CARRIER 1338135052 1338135052 IN IP4 77.72.168.74 s=SIP Call c=IN IP4 77.72.168.74 t=0 0 m=audio 18456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 - --- (11 headers 8 lines) --- Found RTP audio format 0 Found audio description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 77.72.168.74:18456 Peer doesn't provide video list_route: hop: sip:0336@111.111.1.111:5060 set_destination: Parsing sip:0336@111.111.1.111:5060 for address/port to send to set_destination: set destination to 111.111.1.111, port 5060 Transmitting (no NAT) to 111.111.1.111:5060: ACK sip:0336@111.111.1.111:5060 SIP/2.0 Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK5afa4cc0;rport Max-Forwards: 70 From: TOOTAi sip:00333@222.222.22.22;tag=as52190c5c To: sip:0336@111.111.1.111;tag=4e0313ac670313ac4f9920c3173f554 Contact: sip:00333@222.222.22.22 Call-ID: 2c974a0a2b08abe320ed388433e47d7e@222.222.22.22 CSeq: 102 ACK User-Agent: TOOTAiAudio Content-Length: 0 --- -- SIP/myPeerDef-0007 answered SIP/104-0006 Scheduling destruction of SIP dialog '2c974a0a2b08abe320ed388433e47d7e@222.222.22.22' in 32000 ms (Method: INVITE) set_destination: Parsing sip:0336@111.111.1.111:5060 for address/port to send to set_destination: set destination to 111.111.1.111, port 5060 Reliably Transmitting (no NAT) to 111.111.1.111:5060: BYE
[asterisk-users] Which combination of codecs are required?
Hi; In Voicemail.conf If I am using format=h263|gsm ,and i want to store only audio , then it is not storing.In log it shows that video is deposite less then 5 second. If i want to store video and audio both then it will store properly. If am using format=gsm|h263 ,then my Xlite softphone will go to haung. I just want to store audio and video both or some time only audio . 1)Plz guide me which combination of codec will be usefull. 2)Is there is any serial number signifance in format,ie one time if i use as format=h263|gsm and second time i am using format=gsm|h263,why is diffrence come? Thanks Durgesh Mishra Rancore Technologies.-- _ -- 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
